UCM630x Series - User Manual

Thank you for purchasing the Grandstream UCM630X series IP PBX appliance. The UCM6300 series allows businesses to build powerful and scalable

unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized
network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms, and more.

The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from

the desktop, mobile, GVC series devices, and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of

an on-premises IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop and mobile, which

provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300
series also offers cloud setup and management through GDMS and an API for integration with third-party platforms. By offering a high-end unified

communications and collaboration solution packed with a suite of mobility, security, meeting, and collaboration tools, the UCM6300 series provides a

powerful platform for any organization.

Caution

Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User

Manual, could void your manufacturer warranty.

Warning

Please do not use a different power adaptor with the UCM630X as it may cause damage to the product and void the manufacturer warranty.

Alert

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of
Grandstream Networks, Inc. is not permitted.

PRODUCT OVERVIEW

Technical Specifications
The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and

upgrade/provisioning settings for the UCM630X series.

Interfaces

● UCM6301: 1 port with lifeline support

● UCM6302: 2 ports with lifeline support

Analog Telephone FXS Ports ● UCM6304: 4 ports with lifeline support

● UCM6308: 8 ports with lifeline support

Each port supports 3 RN

PSTN Line FXO Ports

● UCM6301: 1 port

● UCM6302: 2 ports

● UCM6304: 4 ports

● UCM6308: 8 ports
 All ports have lifeline capability in case of a power outage.

Network Interfaces Three self-adaptive Gigabit ports (switched, routed, or dual card mode) with PoE+

NAT Router Yes (supports router mode and switch mode)

● UCM6301: USB 3.0, and SD card interface

Peripheral Ports ● UCM6302: USB 2.0, USB 3.0, and SD card interface

● UCM6304/6308: 2x USB 3.0 and SD card interface

● UCM6301/UCM6302: None
LED Indicators
● UCM6304/6308: Power 1/2, FXS, FXO, LAN, WAN, Heartbeat

● UCM6301/UCM6302: 320*240 LCD with touch screen for Shortcut Keys and Scroll Bar
LCD Display
● UCM6304/6308: 128x32 dot matrix graphic LCD with DOWN and OK buttons

Reset Switch Yes, long press for factory reset and short press for reboot

Voice/Video Capabilities

Voice-over-Packet LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter

Capabilities Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss

Voice and Fax Codecs Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38

Video Codecs H.264, H.263, H263+, VP8

QoS Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

Signaling and Control

DTMF Methods Inband, RFC4733, and SIP INFO

Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP
Provisioning Protocol and
endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between the local and remote
Plug-and-Play
trunk

TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS,
Network Protocols
LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN®

API Full API available for third-party platform and application integration

Disconnect Methods Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect

Security
Media Encryption SRTP, TLS1.2, HTTPS, SSH, 802.1x

Physical

● UCM6301/6302: Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A

Universal Power Supply
● UCM6304/6308: Input: 100~240VAC,50/60Hz; Output: DC+12V, 2A

● UCM6301/6302: 270mm(L) x 175mm(W) x 36mm(H)

Dimensions
● UCM6304/6308: 485mm(L) x 187.2mm(W) x 46.2mm(H)

● Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)

Environmental
● Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)

● Wall mount (Unit will be fixed on the wall using screws) & Desktop for UCM6301/6302.
Mounting
● Desktop & Rack mount for UCM6304/6308.

● UCM6301: Unit weight 715g, Package weight 1211g

● UCM6302: Unit weight 725g, Package weight 1221g

Weight
● UCM6304: Unit weight 2490g, Package weight 3260g

● UCM6308: Unit weight 2550g, Package weight 3320g

Additional Features

● Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian,
Polish, Czech, Turkish
Multi-language Support ● Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch,
Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Netherlands

● Customizable language pack to support any other languages

Caller ID Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT

Polarity Reversal/ Wink Yes, with enable/disable option upon call establishment and termination

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/workload, in-queue
Call Center
announcement

Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Telephony Operating System Based on Asterisk version 16

Maximum Call Capacity UCM6301:

● Users: 500

● Concurrent calls (G.711): 75

 ● Max concurrent SRTP calls (G.711): 50

UCM6302:

● Users: 1000

● Concurrent calls (G.711): 150

● Max concurrent SRTP calls (G.711): 100

UCM6304:

● Users: 2000

● Concurrent calls (G.711): 300

● Max concurrent SRTP calls (G.711): 200

UCM6308:

● Users: 3000

● Concurrent calls (G.711): 450

● Max concurrent SRTP calls (G.711): 300

UCM6301:

● 4 Video Conference rooms and up to 20 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen
sharing)

● Voice Conference: Up to 75 parties with G.711

UCM6302:

● 6 Video Conference rooms and up to 30 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen
sharing)

● Voice Conference: Up to 150 parties with G.711

Maximum Attendees of

Conference Bridges UCM6304:

● 8 Video Conference rooms and up to 60 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen
sharing)

● Voice Conference: Up to 200 parties with G.711

UCM6308:

● 10 Video Conference rooms and up to 80 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen
sharing)

● Voice Conference: Up to 300 parties with G.711

Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call

routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group,

Call Features paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, speed dial, call back, dial by name, emergency call,

call follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/ call

completion, voice control

Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the
Wave Mobile App
UCM6300

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system,
Firmware Upgrade
It provides a centralized interface to provision, manage, monitor, and troubleshoot Grandstream products
 ● FCC: Part 15 (CFR 47) Class B, Part 68

● CE: EN 55032, EN 55035, EN61000-3-2, EN61000-3-3, EN 62368.1, ES 203 021, ITU K.21
Compliance ● IC: ICES-003, CS-03 Part I Issue 9

● RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2

● Power adapter: UL 60950-1 or UL 62368-1

Table 1: Technical Specifications

UCM630X FXS ports lifeline functionality: The UCM630X FXS interfaces are metallic through to the FXO interfaces. If there is a power outage, the FXS1 port
will fail over to FXO 1 port, FXS 2 port will fail over to FXO 2 port. The user can still access the PSTN connected with the FXO interfaces from FXS interfaces.

INSTALLATION
Before deploying and configuring the UCM630X series, the device needs to be properly powered up and connected to a network. This section describes

detailed information on the installation, connection, and warranty policy of the UCM630X series.

Equipment Packaging
Main Case 1

Power Adaptor 1

Ethernet Cable 1

Quick Installation Guide 1

Table 2: UCM630X Equipment Packaging

Connect Your UCM630X (UCM6301 as an example)

UCM6301 Top View

 UCM6301 Back View

UCM6301 Front View

To set up the UCM6301, follow the steps below:

1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6301.

2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.

3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6301. Insert the main plug of the power adapter into a surge-

protected power outlet.

4. Wait for the UCM6301 to boot up. The LCD in the front will show the device hardware information when the boot process is done.

5. Once the UCM6301 is successfully connected to the network, press the Home button to display the IP address.

6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.

The ground screw needs to be connected.

The same steps will be used to connect UCM6302/6304/6308.

UCM6302 front and back view

UCM6302 Back View

UCM6302 Front View

UCM6304 front and back view

UCM6304 Front View

UCM6304 Back View

UCM6308 front and back view

UCM6308 Front View

UCM6308 Back View

Safety Compliances

The UCM630X series IP PBX complies with FCC/CE and various safety standards. The UCM630X power adapter is compliant with the UL standard. Use the

universal power adapter provided with the UCM630X package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported

power adapters.

Warranty

If the UCM630X series IP PBX was purchased from a reseller, please contact the company where the device was purchased for replacement, repair, or refund. If

the device was purchased directly from Grandstream, contact our Technical Support Team for an RMA (Return Materials Authorization) number before the

product is returned. Grandstream reserves the right to remedy the warranty policy without prior notification.

Warning

Use the power adapter provided with the UCM630X series IP PBX. Do not use a different power adapter as this may damage the device. This type of damage is

not covered under warranty.

GETTING STARTED
To get started with the UCM630X setup process, use the following available interfaces: LCD display, and web portal.

The LCD display shows hardware, software, interface status, and network information and can be navigated via the Slide control and Touch keys. From
here, users can configure basic network settings, run diagnostic tests, and factory reset.
 The web portal (may also be referred to as web UI in this guide) is the primary method of configuring the UCM.

This section will provide step-by-step instructions on how to use these interfaces to quickly set up the UCM and start making and receiving calls with it.

Use the LCD Menu

Idle Screen

Once the device has booted up completely, the LCD will show the UCM model, hardware version, and IP address. Upon menu key press timeout (30

seconds), the screen will default back to this information.

Menu

Pressing the Home button will show the main menu. All available menu options are found in [Table 3: LCD Menu Options].

Menu Navigation

Scrolling down using slide control through the menu options. Press the OK button to select an option.

Exit

Selecting the Back option will return to the previous menu. For the Device Info, Network Info, and Web Info screens that have no Back option, pressing the
OK button will return to the previous menu.

LCD Backlight

The LCD backlight will turn on upon button press and will go off when idle for 30 seconds.

The following table summarizes the layout of the LCD menu of UCM630x.

● Critical Events
View Events
● Other Events

● Hardware: Hardware version number

● Software: Software version number

● P/N: Part number

Device Info
● WAN MAC: WAN side MAC address

● LAN MAC: LAN side MAC address

● Uptime: System uptime

● WAN Mode: DHCP, Static IP, or PPPoE

● WAN IP: IP address

Network Info ● WAN Subnet Mask

● LAN IP: IP address

● LAN Subnet Mask

Network Menu
● WAN Mode: Select WAN mode as DHCP, Static IP, or PPPoE
 ● Static Route Reset: Select this to reset static route settings.

● Reboot

● Factory Reset

● LCD Test Patterns: Press DOWN and OK buttons to scroll through and select different LCD patterns to test.
Once a test is done, press the OK button to return to the previous menu.

● Fan Mode: Select Auto or On.

● LED Test Patterns: All On, All Off, and Blinking are the available options. Selecting Back in the menu will
Factory Menu
revert the LED indicators to their actual status.

● RTC Test Patterns: Select either 2022-02-22 22:22 or 2011-01-11 11:11 to start the RTC (Real-Time Clock)
test pattern. Check the system time from either the LCD idle screen or in the web portal System

Status🡪System Information🡪General page. To revert to the correct time, manually reboot the device.

● Hardware Testing: Select Test SVIP to verify hardware connections within the device. The result will display

on the LCD when the test is complete.

● Protocol: Web access protocol (HTTP/ HTTPS). HTTPS is used by default.

Web Info
● Port: Web access port number, which is 8089 by default.

● Enable SSH
SSH Switch ● Disable SSH

SSH access is disabled by default

Use the LED Indicators

The UCM6301/6302 has LED indicators on the network port to display connection status and the following picture shows the other ports’ status.

Ports Status

The UCM6304/6308 has LED indicators in the front to display the connection status. The following table shows the status definitions.
 LED Indicator LED Status

Power 1/Power 2

PoE

LAN
Solid: Connected

WAN Fast Blinking: Data Transferring

USB Slow Blinking: Trying to connect

OFF: Not Connected

SD

FXS ports

FXO ports

Table 4: UCM6304/6308 LED Indicators

Using the Web UI

Accessing the Web UI

The UCM’s web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a web browser

such as Microsoft IE (version 8+), Mozilla Firefox, Google Chrome, etc. To access the UCM’s web portal, follow the steps below:

UCM6302 Web GUI Login Page

1. Make sure your computer is on the same network as the UCM.

2. Make sure that the UCM’s IP address is displayed on its LCD.

3. Enter the UCM’s IP address into a web browsers’ address bar. The login page should appear (please see the above image).

4. Enter default administrator username “admin” and password can be found on the sticker at the back of the UCM.
 5.

By default, the UCM630X has Redirect From Port 80 enabled. As such, if users type in the UCM630X IP address in the web browser, the web page will be

automatically redirected to the page using HTTPS and port 8089. For example, if the LCD shows 192.168.40.167, and 192.168.40.167 is entered into the web

browser, the web page will be redirected to: https://192.168.40.167:8089

The option Redirect From Port 80 can be found under the UCM630X Web GUI🡪System Settings🡪HTTP Server.

Setup Wizard

When you log in to the UCM Web GUI interface for the first time, the system will automatically start the setup wizard and expand the description of the

instant messaging soft terminal supporting system ( Wave). Click “Learn more” to open the Wave client download interface.

UCM630x Web GUI Wave guide interface

The setup wizard guides users to complete basic configuration, such as administrator password modification, network settings, time zone settings, extension

settings, and routing rule configuration, etc.

UCM630X Setup Wizard

The setup wizard can be closed and reopened at any time. At the end of the wizard, a summary of the pending configuration changes can be reviewed before

applying them.
Main Settings

There are 8 main sections in the web portal to manage various features of the UCM.

System Status: Displays the dashboard, system information, current active calls, and network status.

Extensions/Trunks: Manages extensions, trunks, and routing rules.

Call Features: Manages various features of the UCM such as the IVR and voicemail.

PBX Settings: Manages the settings related to PBX functionality such as SIP settings and interface settings.

System Settings: Manages the settings related to the UCM system itself such as network and security settings.

CDR: Contains the call detail records, statistics, and audio recordings of calls processed by the UCM.

Other Features: Manages the settings of features unrelated to core PBX functionality such as Zero Config provisioning and CRM/PMS integrations.

Maintenance: Manages settings and logs related to system management and maintenance such as
user management, activity logs, backup settings, upgrade settings, and troubleshooting tools.

Web GUI Languages

Currently the UCM630X series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian,

Polish, German, Turkish, and Czech.

Users can select the UCM’s web UI display language in the top-right corner of the page.

UCM630X Web GUI Language

Web GUI Search Bar

Users can search for options in the web portal with the search bar on the top right of the page.
 Web GUI Search Bar

Saving and Applying Changes

After making changes to a page, click on the “Save” button to save them and then the “Apply Changes” button that finalizes the changes. If a modification
requires a reboot, a prompt will appear asking to reboot the device.

Setting Up an Extension
Power on the UCM630X and your SIP endpoint. Connect both devices to the same network and follow the steps below to set up an extension.

1. Log into the UCM web portal and navigate to Extension/Trunk🡪Extensions

2. Click on the “Add” button to start creating a new extension. The Extension and SIP/IAX Password information will be used to register this extension. To

set up voicemail, the Voicemail Password will be required.

3. To register an endpoint to this extension, go into your endpoint’s web UI and edit the desired account. Enter the newly created extension’s number, SIP

user ID, and password into their corresponding fields on the endpoint. Enter the UCM’s IP address into the SIP server field. If setting up voicemail, enter

*97 into the Voice Mail Access Number field. This field may be named differently on other devices.

4. To access the extension’s voicemail, use the newly registered extension to dial *97 and access the personal voicemail system. Once prompted, enter the

voicemail password. If successful, you will now be prompted with various voicemail options.

5. You have now set up an extension on an endpoint.

SYSTEM SETTINGS
This section will explain the available system-wide parameters and configuration options on the UCM630X series. This includes settings for the following

items: General Settings, Cloud IM, HTTP server, network Settings, OpenVPN, DDNS Settings, Security Settings, LDAP server, Time settings, Email settings,

and TR-069.

General Settings
System administrators can prevent the UCM from making calls and/or writing to the data partition (e.g., CDR, recordings, etc.) once the system reaches a

specified threshold of storage usage and CPU usage respectively. These options are located in the System Settings 🡪 General Settings page.
 General Settings Interface

General Settings

Device Name Configure the name of the UCM.

Enable CPU Flow

Enables the CPU flow control.
Control

Used to set the threshold generated by the CPU Flow Control. When the system CPU reaches the threshold, it will prohibit the
CPU Flow Control new calls.
Threshold
The default value is 90%.

Data Partition Write Used to set a threshold to stop writing data partition. When the disk data partition reaches the threshold configured, the data
Threshold partition writing will be stopped. The default value is 90%.

Table 5: General Settings Parameters

IM Settings

Cloud IM Service

After enabling Cloud IM, it means that all IM data in Wave is stored in the external server Cloud IM, and is no longer stored locally in UCM.GDMS can

configure the External Cloud IM service for UCM devices. At this time, the UCM device synchronizes the configuration item information.
 Cloud IM Service

Cloud IM Service

If you have purchased the UCM Cloud IM package or purchased the Grandstream IM server, you can configure it.

Enable Cloud IM If you have not purchased it, the configuration will not take effect, but UCM local IM service is allowed. Please

note that after enabling this feature, local chat data will not be visible.

If enabled, the local proxy will be used to forward files and text messages if the IM server cannot be connected to
Local Proxy
upon Wave login due to certificate issues.

The address of the server that provides IM service, you can fill in the address of the Cloud IM server provided by
Cloud IM Server Address
the RemoteConnect package or the IM server address of the GDMS.

Service ID The service ID of the Cloud IM server.

Key The Key to the Cloud IM server.

Company Name Company Name

Trusted User The trusted user of the cloud IM. Only letters, numbers, and special characters are allowed.

Prepend As the extension prefix, it is added before the extension number.

Sync Local Chat Data Syncing existing local chat data to Cloud IM server. The Wave chat feature will not be available during the syncing

process. It is recommended to avoid syncing during active working hours.

- Time Range
 ● All

● Last 12 Months

● Last 6 Months

● Last 3 Months

● Last Month

- Data Type

● IM Data

● Images

● Files

Note

Please note that synchronization of the local chat can only occur in the initial connection to a Cloud IM Server . If the UCM is already connected to a Cloud IM

server, or the Cloud IM server has already been synced to other UCMs, local chat data will not be able to be synced.

Only account details and department information will be synced on local IM and cloud IM. Other configurations such as profile picture, work status, and favorite

contacts will not be synced and these are stored in local IM or cloud IM respectively. Therefore, please be aware that when switching between local IM and cloud

IM, part of the data cannot be synced and the previously stored data on local IM or cloud IM (depending on which one is switched to) will be retrieved.

IM Server

If Enable IM Server Mode is toggled on, UCM will function only as an IM server. The UCM management portal will remove PBX-related services and

supports the binding of multiple cross-region UCM devices. The UCM device that wants to bind the IM server address is also bound by turning on the Cloud

IM mode, and the IM data in his Wave is stored in this IM server.

IM server configuration interface

Company name The entered company name

Server Address The domain name or IP address of the Cloud IM server.

Service ID The service ID of the Cloud IM server.

Key The Key of the Cloud IM server.

 Trusted User The trusted user of the cloud IM. Only letters, numbers, and special characters are allowed.

Bound Device Information

Department The department represented by the bound UCM.

MAC Address MAC address of the bound UCM device.

Dial Prefix Extension prefix

HTTP Server
The UCM630X’s embedded web server responds to HTTPS GET/POST requests and allows users to configure the UCM via web browsers such as Microsoft
IE, Mozilla Firefox, and Google Chrome. By default, users can access the UCM by just typing its IP address into a browser address bar. The browser will
automatically be redirected to HTTPS using port 8089. For example, typing in “192.168.40.50” into the address bar will redirect the browser to

“https://192.168.40.50:8089”. This behavior can be changed in the System Settings🡪HTTP Server page.

Toggles automatic redirection to UCM’s web portal from port 80. If disabled, users will need to manually add the

Redirect From Port 80 UCM’s configured HTTPS port to the server address when accessing the UCM web portal via browser. Default is

“Enabled”.

Port Specifies the port number used to access the UCM HTTP server. Default is “8089”.

If enabled, only the server addresses in the whitelist will be able to access the UCM’s web portal. It is highly

Enable IP Address Whitelist recommended to add the IP address currently used to access the UCM web page before enabling this option. Default is

“Disabled”.

List of addresses that can access the UCM web portal.

Permitted IP(s)
Ex: 192.168.6.233 / 255.255.255.255

Configure a URL and port (optional) used to access the UCM web portal or a public link to the video conference room if
External Host
the UCM is behind NAT.

Wave Settings

Configure a URL and port (optional) used to access the UCM web portal or a public link to the video conference room if
External Host
the UCM is behind NAT.

Port The port to access Wave Web and Wave Mobile. If behind NAT, please make sure to map the external port to this port.

Certificate Settings

If enabled, the default browser certificate will be automatically renewed after 398 days (the max certificate validity
Default Certificate Auto Renewal
period of Chrome, Firefox, and Safari browsers). User-defined certificates are not affected.

Options Selects the method of acquiring SSL certificates for the UCM web server. Two methods are currently available:
 ● Upload Certificate: Upload the appropriate files from one’s own PC.

● Request Certificate: Enter the domain for which to request a certificate for from “Let’s Encrypt”.

Uploads the private key for the HTTP server.

TLS Private Key Note: Key file must be under 2MB in file size and *.pem format. The file name will automatically be changed to

“private.pem”.

Uploads the certificate for the HTTP server.

TLS Cert Note: Certificate must be under 2MB in file size and *.pem format. This will be used for TLS connections and contains

a private key for the client and a signed certificate for the server.

Enter the domain to request the certificate for and click on "Request Certificate" button.
Domain
Request Certificate

If the protocol or port has been changed, the user will be logged out and redirected to the new URL.

Network Settings
After successfully connecting the UCM630X to the network for the first time, users could log in to the Web GUI and go to System Settings🡪Network
Settings to configure the network parameters for the device.

UCM630X supports Route/Switch/Dual mode functions.

In this section, all the available network setting options are listed for all models. Select each tab in Web GUI🡪System Settings🡪Network Settings page to
configure LAN settings, WAN settings, 802.1X, and Port Forwarding.

Basic Settings

Please refer to the following tables for basic network configuration parameters on UCM6301, UCM6302, UCM6304, and UCM6308, respectively.

Select "Route", "Switch" or "Dual" mode on the network interface of UCM630X. The default setting is "Switch".

● Route: WAN port will be used for the uplink connection. LAN port will function similarly to a regular router
port.
Method ● Switch: WAN port will be used for the uplink connection. LAN port will be used as a bridge for connections.

● Dual: Both WAN and LAN ports will be used for uplink connections labeled as LAN2 and LAN1, respectively.

The port selected as the Default Interface will need to have a gateway IP address configured if it is using a

static IP.

MTU Specifies the maximum transmission unit value. Default is 1492.

IPv4 Address

Preferred DNS Server If configured, this will be used as the Primary DNS server.

WAN (when "Method" is set to "Route")

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1 Enter the DNS server 1 address for static IP settings.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username to connect via PPPoE.

Password Enter the password to connect via PPPoE.

Layer 2 QoS 802.1Q/VLAN Tag Assign the VLAN tag of the layer 2 QoS packets for the WAN port. The default value is 0.

Layer 2 QoS 802.1p Priority Value Assign the priority value of the layer 2 QoS packets for the WAN port. The default value is 0.

LAN (when Method is set to "Route")

IP Address Enter the IP address assigned to the LAN port. The default setting is 192.168.2.1.

Subnet Mask Enter the subnet mask. The default setting is 255.255.255.0.

DHCP Server Enable Enable or disable DHCP server capability. The default setting is "Yes".

DNS Server 1 Enter DNS server address 1. The default setting is 8.8.8.8.

DNS Server 2 Enter DNS server address 2. The default setting is 208.67.222.222.

Allow IP Address From Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.

Allow IP Address To Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.

Default IP Lease Time Enter the IP lease time (in seconds). The default setting is 43200.

LAN (when Method is set to "Switch")

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.
Gateway IP Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1 Enter the DNS server 1 address for static IP settings.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username to connect via PPPoE.

Password Enter the password to connect via PPPoE.

Layer 2 QoS 802.1Q/VLAN Tag Assign the VLAN tag of the layer 2 QoS packets for the LAN port. The default value is 0.

Layer 2 QoS 802.1p Priority Value Assign the priority value of the layer 2 QoS packets for the LAN port. The default value is 0.

LAN 1 / LAN 2 (when Method is set to "Dual")

If "Dual" is selected as "Method", users will need to assign the default interface to be LAN 1 (mapped to

Default Interface UCM6302 WAN port) or LAN 2 (mapped to UCM6302 LAN port) and then configure network settings for LAN

1/LAN 2. The default interface is LAN 2.

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Enter the gateway IP address for static IP settings when the port is assigned as the default interface. The default
Gateway IP
setting is 0.0.0.0.

DNS Server 1 Enter the DNS server 1 address for static IP settings.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username to connect via PPPoE.

Password Enter the password to connect via PPPoE.

Assign the VLAN tag of the layer 2 QoS packets for the LAN port.
Layer 2 QoS 802.1Q/VLAN Tag
The default value is 0.

Layer 2 QoS 802.1p Priority Value Assign the priority value of the layer 2 QoS packets for the LAN port. The default value is 0.

IPv6 Address

WAN (when "Method" is set to "Route")

IP Method Select Auto or Static. The default setting is Auto

IP Address Enter the IP address for static IP settings.

IP Prefixlen Enter the Prefix length for static settings. Default is 64

DNS Server 1 Enter the DNS server 1 address for static settings.

DNS Server 2 Enter the DNS server 2 address for static settings.

LAN (when Method is set to "Route")

Select Disable, Auto, or DHCPv6.

● Disable: the DHCPv6 server is disabled.

DHCP Server ● Auto: Stateless address auto configuration using NDP protocol.

● DHCPv6: Stateful address auto configuration using DHCPv6 protocol.

The default setting is Disabled.

DHCP Prefix Enter DHCP prefix. (Default is 2001:db8:2:2::)

DHCP prefixlen Enter the Prefix length for static settings. Default is 64

DNS Server 1 Enter the DNS server 1 address for static settings. Default is (2001:4860:4860::8888 )

DNS Server 2 Enter the DNS server 2 address for static settings. Default is (2001:4860:4860::8844 )

Allow IP Address From Configure starting IP address assigned by the DHCP prefix and DHCP prefixlen.

Allow IP Address To Configure the ending IP address assigned by the DHCP Prefix and DHCP prefixlen.

Default IP Lease Time Configure the lease time (in second) of the IP address.

LAN (when Method is set to "Switch")

IP Method Select Auto or Static. The default setting is Auto

IP Address Enter the IP address for static IP settings.

IP Prefixlen Enter the Prefix length for static settings. Default is 64

DNS Server 1 Enter the DNS server 1 address for static settings.

DNS Server 2 Enter the DNS server 2 address for static settings.

LAN 1 / LAN 2 (when Method is set to "Dual")

Users will need to assign the default interface to be LAN 1 (mapped to UCM630X WAN port) or LAN 2 (mapped
Default Interface
to UCM630X LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 1.
IP Method Select Auto or Static. The default setting is Auto

IP Address Enter the IP address for static IP settings.

IP Prefixlen Enter the Prefix length for static settings. Default is 64

DNS Server 1 Enter the DNS server 1 address for static settings.

DNS Server 2 Enter the DNS server 2 address for static settings.

Network Port Traffic Control

LAN (when Method is set to "Switch")

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that

Enable Network Port Traffic Storm impacts the overall performance of the network.

Alert Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System

Events.

You can monitor the traffic in the RX direction on each network port and generate an alarm when the

Network Port Receiving Traffic Control corresponding alarm event is turned on and the set threshold value is exceeded.

The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN 1 & LAN 2 (when Method is set to "Dual")

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that

Enable Network Port Traffic Storm impacts the overall performance of the network.

Alert Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System

Events.

LAN1 & LAN2 You can monitor the traffic in the RX direction on each network port and generate an alarm when the

- Network Port Receiving Traffic corresponding alarm event is turned on and the set threshold value is exceeded.

Control The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN & WAN (When Method is set to Route Mode)

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that

Enable Network Port Traffic Storm impacts the overall performance of the network.

Alert Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System

Events.

You can monitor the traffic in the RX direction on each network port and generate an alarm when the
WAN:
corresponding alarm event is turned on and the set threshold value is exceeded.
Network Port Receiving Traffic Control
The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN: You can monitor the traffic in the RX direction on each network port and generate an alarm when the

Network Port Receiving Traffic Control corresponding alarm event is turned on and the set threshold value is exceeded.
 The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

Method: Route

When the UCM630X has the method set to Route in network settings, WAN port interface is used for uplink connection and LAN port interface is used as a
router. Please see a sample diagram below.

UCM6302 Network Interface Method: Route

Method: Switch

WAN port interface is used for uplink connection; LAN port interface is used as a room for PC connection.

UCM6302 Network Interface Method: Switch

Method: Dual

Both WAN port and LAN port are used for the uplink connection. Users will need to assign LAN 1 or LAN 2 as the default interface in option “Default

Interface” and configure “Gateway IP” if static IP is used for this interface.
 UCM6302 Network Interface Method: Dual

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to a device before the device can access the

Internet or other LAN resources. The UCM630X supports 802.1X as a supplicant/client to be authenticated.

The following diagram and figure show UCM630X uses 802.1X mode “EAP-MD5” on the WAN port as a client in the network to access the Internet.

UCM630X Using 802.1X as Client

UCM630X Using 802.1X EAP-MD5

The following table shows the configuration parameters for 802.1X on UCM630X. Identity and MD5 password are required for authentication, which should

be provided by the network administrator obtained from the RADIUS server. If “EAP-TLS” or “EAP-PEAPv0/MSCHAPv2” is used, users will also need to
upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.
 Select 802.1X mode. The default setting is “Disable”. The supported 802.1X modes are:

EAP-MD5
802.1X Mode
EAP-TLS

EAP-PEAPv0/MSCHAPv2

Identity Enter 802.1X mode Identity information.

MD5 Password Enter 802.1X mode MD5 password information.

802.1X CA Certificate Select 802.1X certificate from local PC and then upload.

802.1X Client Certificate Select 802.1X client certificate from local PC and then upload.

Table 10: UCM630X Network Settings🡪802.1X

Static Routes

The UCM630X provides users static routing capability that allows the device to use manually configured routes, rather than information only from dynamic

routing or gateway configured in the UCM630X Web GUI🡪System Settings🡪Network Settings🡪Basic Settings to forward traffic. It can be used to define
a route when no other routes are available or necessary or used in complementary with existing routing on the UCM630X as a failover backup, etc.

Click on “Add IPv4 Static Route” to create a new IPv4 static route or click ”Add IPv6 Static Route” to create a new IPv6 static route. The

configuration parameters are listed in the table below.

Once added, users can select to edit the static route.

Select to delete the static route.

Configure the destination IPv4 address or the destination IPv6 subnet for the UCM630X to reach using the static route.

Example:
Destination
IPv4 address – 192.168.66.4

IPv6 subnet – 2001:740:D::1/64

Configure the subnet mask for the above destination address. If left blank, the default value is 255.255.255.255.

Subnet
Example:
Mask

255.255.255.0

Configure the IPv4 or IPv6 gateway address so that the UCM630X can reach the destination via this gateway. The gateway address is

optional.

Gateway
Example:

192.168.40.5 or 2001:740:D::1
 Specify the network interface on the UCM630X to reach the destination using the static route.
Interface
LAN interface is eth0; WAN interface is eth1.

Table 11: UCM630X Network Settings🡪Static Routes

Static routes configuration can be reset from the LCD menu🡪Network Menu.

The following diagram shows a sample application of static route usage on UCM6304.

UCM630X Network Settings🡪Static Routes

The network topology of the above diagram is as below:

Network 192.168.69.0 has IP phones registered to UCM6304 LAN 1 address

Network 192.168.40.0 has IP phones registered to UCM6304 LAN 2 address

Network 192.168.66.0 has IP phones registered to UCM6304 via VPN

Network 192.168.40.0 has a VPN connection established with network 192.168.66.0

In this network, by default, the IP phones in network 192.168.69.0 are unable to call IP phones in network 192.168.66.0 when registered on different

interfaces on the UCM6304. Therefore, we need to configure a static route on the UCM6304 so that the phones in isolated networks can make calls between

each other.
 UCM6304 Static Route Configuration

Port Forwarding

The UCM network interface supports the router function which provides users the ability to do port forwarding. If LAN mode is set to “Route” under Web

GUI🡪System Settings🡪Network Settings🡪Basic Settings page, port forwarding is available for configuration.

The port forwarding configuration is under Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding page. Please see related settings in the table
below.

Specify the WAN port number or a range of WAN ports. An unlimited number of ports can be configured.

Note:

WAN Port When it is set to a range, the WAN port, and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN

port: 1000-1005, and access from the WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000

will be port forwarding to LAN port 1000.

LAN IP Specify the LAN IP address.

Specify the LAN port number or a range of LAN ports.

Note:
LAN Port
When it is set to a range, the WAN port, and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN

port: 1000-1005, and access from the WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000

will be port forwarding to LAN port 1000.

Protocol Type Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in the selected port. The default setting is "UDP Only".

Table 12: UCM630X Network Settings🡪Port Forwarding

The following figures demonstrate a port forwarding example to provide a phone’s Web GUI access to the public side.

UCM630X network mode is set to “Route”.

UCM630X WAN port is connected to the uplink switch, with a public IP address configured, e.g. 1.1.1.1.

UCM630X LAN port provides a DHCP pool that connects to multiple phone devices in the LAN network 192.168.2.x. The UCM60X is used as a router,
with gateway address 192.168.2.1.

There is a GXP2160 connected under the LAN interface network of the UCM630X. It obtains IP address 192.168.2.100 from the UCM630X DHCP pool.

On the UCM630X Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding, configure a port forwarding

entry as the figure shows below.

 WAN Port: This is the port opened on the WAN side for access purposes.

LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM630X.

LAN Port: This is the port opened on the GXP2160 side for access purposes.

Protocol Type: We select TCP here for Web GUI access using HTTP.

Create New Port Forwarding

UCM630X Port Forwarding Configuration

This will allow users to access the GXP2160 Web GUI from the public side, by typing in the public IP address (example: 1.1.1.1:8088).

GXP2160 Web Access using UCM6302 Port Forwarding

ARP Settings

The ARP settings can be configured under Web GUI🡪System Settings🡪Network Settings🡪ARP Settings

ARP GC A minimum number of entries to keep. The garbage collector will not purge entries if there are fewer than this number. The default value
Threshold 1 is 128.

ARP GC Threshold when garbage collector becomes more aggressive about purging entries. Entries older than 5 seconds will be cleared when over

Threshold 2 this number. The default value is 512.

ARP GC The maximum number of non-PERMANENT neighbor entries allowed. Increase this when using large numbers of interfaces and when

Threshold 3 communicating with large numbers of directly connected peers. The default value is 1024.

Table 13: ARP Settings

OpenVPN®
OpenVPN® settings allow the users to configure UCM630X to use VPN features, the following table gives details about the various options to configure the

UCM as OpenVPN Client.

Open VPN® Feature on the UCM630X

OpenVPN® Enable Enable / Disable the OpenVPN® feature.

Configuration Method Select the OpenVPN® configuration method.

Manual Configuration: Allows to configure OpenVPN® settings manually.

 Upload Configuration File: Allows to upload .ovpn and .conf files to the UCM and to automatically configure

OpenVPN® settings.

OpenVPN® Server Address Configures the hostname/IP and port of the server. For example 192.168.1.2:22

OpenVPN® Server Protocol Specify the protocol user, user should use the same settings as used on the server

Use the same setting as used on the server.

OpenVPN® Device mode ● Dev TUN: Create a routed IP tunnel.

● Dev TAP: Create an Ethernet tunnel.

Compress tunnel packets using the LZO algorithm on the VPN link. Do not enable this unless it is also enabled in the
OpenVPN® Use Compression
server config file.

Enable Weak SSL Ciphers Either to enable the Weak SSL ciphers or not.

OpenVPN® Encryption Specify the cryptographic cipher. Users should make sure to use the same setting that they are using on the OpenVPN

Algorithm server.

OpenVPN® CA Cert Upload as SSL/TLS root certificate. This file will be renamed as ‘ca.crt’ automatically.

OpenVPN® Client Cert Upload a client certificate. This file will be renamed as ‘client.crt’ automatically.

OpenVPN® Client Key Upload a client private key. This file will be renamed as ‘client.key’ automatically.

Username Username used to authenticate into the server.

Password Password used to authenticate into the server.

DDNS Settings
DDNS setting allows users to access UCM630X via domain name instead of IP address.

The UCM supports DDNS service from the following DDNS provider:

dydns.org

noip.com

freedns.afraid.org

zoneedit.com

oray.net

Here is an example of using noip.com for DDNS.

1. Register domain in DDNS service provider. Please note the UCM630X needs to have public IP access.
 Register Domain Name on noip.com

2. On Web GUI🡪System Settings🡪Network Settings🡪DDNS Settings, enable DDNS service and configure username, password, and host name.

UCM630X DDNS Setting

3. Now you can use a domain name instead of an IP address to connect to the UCM630X Web GUI.

Using Domain Name to Connect to UCM630X

Security Settings
The UCM630X provides users firewall security configurations to prevent certain malicious attacks to the UCM630X system. Users could configure to allow,
restrict, or reject specific traffic through the device for security and bandwidth purpose. The UCM630X also provides the Fail2ban feature for authentication

errors in SIP REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM630X, go to Web GUI🡪System Settings🡪Security Settings

page.

Static Defense

Under Web GUI🡪System Settings🡪Security Settings🡪Static Defense page, users will see the following information:

Current service information with port, process, and type.

Typical firewall settings.

Custom firewall settings.

The following table shows a sample current service status running on the UCM630X.

Port Process Type Protocol or Service

7777 Asterisk TCP/IPv4 SIP

389 Slapd TCP/IPv4 LDAP

6060 zero_config UDP/IPv4 UCM630X zero_config service

5060 Asterisk UDP/IPv4 SIP

4569 Asterisk UDP/IPv4 SIP

38563 Asterisk udp/ipv4 SIP

10000 gs_avs udp/ipv4 gs_avs

10001 gs_avs udp/ipv4 gs_avs

10002 gs_avs udp/ipv4 gs_avs

10003 gs_avs udp/ipv4 gs_avs

10004 gs_avs udp/ipv4 gs_avs

10005 gs_avs udp/ipv4 gs_avs

10006 gs_avs udp/ipv4 gs_avs

10007 gs_avs udp/ipv4 gs_avs

 10010 gs_avs udp/ipv4 gs_avs

10012 gs_avs udp/ipv4 gs_avs

10013 gs_avs udp/ipv4 gs_avs

10014 gs_avs udp/ipv4 gs_avs

10015 gs_avs udp/ipv4 gs_avs

10018 gs_avs udp/ipv4 gs_avs

10019 gs_avs udp/ipv4 gs_avs

10020 gs_avs udp/ipv4 gs_avs

6066 Python udp/ipv4 python

3306 Mysqld tcp/ipv4 mysqld

45678 Python udp/ipv4 python

8439 Lighttpd tcp/ipv4 HTTP

8088 asterisk tcp/ipv4 SIP

8888 Pbxmid tcp/ipv4 pbxmid

25 Master tcp/ipv4 master

636 Slapd tcp/ipv4 SLDAP

4569 asterisk udp/ipv6 SIP

42050 asterisk udp/ipv6 SIP

7681 Pbxmid tcp/ipv4 pbxmid

Table 15: UCM630X Firewall🡪Static Defense🡪Current Service

For typical firewall settings, users could configure the following options on the UCM630X.

Ping Defense If enabled, ICMP response will not be allowed for Ping requests. The default setting is disabled. To enable or disable it, click on the

Enable check box for the LAN or WAN (UCM630X) interface.

 Allows the UCM630X to handle excessive amounts of SYN packets from one source and keep the web portal accessible. There are
two options available and only one of these options may be enabled at one time.

SYN-Flood eth(0)LAN defends against attacks directed to the LAN IP address of the UCM630X.

Defense Enable eth(1)WAN defends against attacks directed to the WAN IP address of the UCM630X.

SYN Flood Defense will limit the amount of SYN packets accepted by the UCM from one source to 10 packets per second. Any

excess packets from that source will be discarded.

Ping-of-Death Enable to prevent Ping-of-Death attack to the device. The default setting is disabled. To enable or disable it, click on the check box
Defense Enable for the LAN or WAN (UCM630X) interface.

Table 16: Typical Firewall Settings

Under “Custom Firewall Settings”, users could create new rules to accept, reject or drop certain traffic going through the UCM630X. To create a new rule,
click on the “Create New Rule” button and a new window will pop up for users to specify rule options.

Right next to the “Create New Rule” button, there is a checkbox for the option “Reject Rules”. If it is checked, all the rules will be rejected except the firewall

rules listed below. In the firewall rules, only when there is a rule that meets all the following requirements, the option “Reject Rules” will be allowed to check:

Action: “Accept”

Type “In”

The destination port is set to the system login port (e.g., by default 8089)

The protocol is not UDP

Create New Firewall Rule

Rule Name Specify the Firewall rule name to identify the firewall rule.

Select the action for the Firewall to perform.

ACCEPT
Action
REJECT

DROP
 Select the traffic type.

IN
Type
If selected, users will need to specify the network interface “LAN” or “WAN” (for UCM630X) for the incoming traffic.

OUT

Interface Select the interface to use the Firewall rule.

Select the service type.

FTP

SSH

Telnet

Service HTTP

LDAP

Custom

If “Custom” is selected, users will need to specify Source (IP and port), Destination (IP and port), and Protocol (TCP, UDP, or Both)
for the service. Please note if the source or the destination field is left blank, it will be used as “Anywhere”.

Source IP Address Configure a source subnet and port. If set to “Anywhere” or left empty, traffic from all addresses and ports will be accepted. A single

and Port port or a range of ports can be specified (e.g., 10000, 10000-20000).

Destination IP Configure a destination subnet and port. If set to “Anywhere” or left empty, traffic can be sent to all addresses and ports. A single
Address and Port port or a range of ports can be specified (e.g., 10000, 10000-20000).

Protocol Select the protocol for the rule to be used.

Table 17: Firewall Rule Settings

Save the change and click on the “Apply” button. Then submit the configuration by clicking on “Apply Changes” on the upper right of the web page. The new

rule will be listed at the bottom of the page with sequence number, rule name, action, protocol, type, source, destination, and operation. More operations are

below:

Click on to edit the rule.

Click on to delete the rule.

Dynamic Defense

Dynamic defense is supported on the UCM630X series. It can blacklist hosts dynamically when the LAN mode is set to “Route” under Web GUI🡪System

Settings🡪Network Settings🡪Basic Settings page. If enabled, the traffic coming into the UCM630X can be monitored, which helps prevent massive

connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM630X firewall, which will then be displayed on

the web page. Please refer to the following table for dynamic defense options on the UCM630X.
 Dynamic Defense
Enable dynamic defense. The default setting is disabled.
Enable

Blacklist Update
Configure the blacklist update time interval (in seconds). The default setting is 120.
Interval

Connection Configure the connection threshold. Once the number of connections from the same host reaches the threshold, it will be added to

Threshold the blacklist. The default setting is 100.

Allowed IPs and ports range, multiple IP addresses, and port range.

Dynamic Defense
For example:
Whitelist

192.168.2.100-192.168.2.105, 1000:9999

Table 18: UCM630X Firewall Dynamic Defense

The following figure shows a configuration example like this:

If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM630X it will be added to the UCM630X blacklist.

This host 192.168.5.7 will be blocked by the UCM630X for 500 seconds.

Since IP range 192.168.5.100-192.168.5.200 is in the whitelist if a host initiates more than 20 TCP connections to the UCM630X it will not be added to
the UCM630X blacklist. It can still establish a TCP connection with the UCM630X.

Configure Dynamic Defense

Fail2ban

Fail2Ban feature on the UCM630X provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. Once

the entry is detected within “Max Retry Duration”, the UCM630X will act to forbid the host for a certain period as defined in “Banned Duration”. This feature
helps prevent SIP brute force attacks on the PBX system.
 Fail2ban Settings

Global Settings

Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and "Asterisk Service"
Enable Fail2Ban
are turned on to use Fail2Ban for SIP authentication on the UCM630X.

Configure the duration (in seconds) for the detected host to be banned. The default setting is 600. If set to 0, the
Banned Duration
host will be always banned.

Within this duration (in seconds), if a host exceeds the max times of retry as defined in "MaxRetry", the host will
Max Retry Duration
be banned. The default setting is 600.

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The
MaxRetry
default setting is 5.

Configure IP address, CIDR mask, or DNS host in the whitelist. Fail2Ban will not ban the host with a matching

Fail2Ban Whitelist address in this list. Up to 20 addresses can be added to the list descriptions/comments can be added for each

whitelist entry for admin to log what’s the whitelist IP address is for.

Local Settings

Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and
Asterisk Service
"Asterisk Service" are turned on to use Fail2Ban for SIP authentication on the UCM630X.

Configure the listening port number for the service. By default, port 5060 will be used for UDP and TCP, and port
Listening Port Number
5061 will be used for TCP.

MaxRetry Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The

default setting is 5. Please make sure this option is properly configured as it will override the "MaxRetry" value
 under "Global Settings".

Enables defense against excessive login attacks to the UCM’s web GUI.
Login Attack Defense
The default setting is disabled.

This is the Web GUI listening port number which is configured under System Settings🡪 HTTP Server🡪 Port.
Listening Port Number
The default is 8089.

When the number of failed login attempts from an IP address exceeds the MaxRetry number, that IP address will
MaxRetry
be banned from accessing the Web GUI.

Customer Service System Call Defense Enable call defense in the customer service system. Off by default.

The current service listening port. Default UDP port: 5060, TCP port: 5060, 5061, WebSocket communication port:
Listening Port Number
8088.

Set the maximum number of calls allowed in the "time span". The local matching threshold has a higher priority
MaxRetry
than the global matching threshold. The default setting is 5.

Blacklist

Blacklist Users will be able to view the IPs that have been blocked by UCM.

SSH Access

SSH switch now is available via Web GUI and LCD. Users can enable or disable SSH access directly from Web GUI or LCD screen. For web SSH access,

please log in to UCM630X web interface and go to Web GUI🡪System Settings🡪Security Settings🡪SSH Access.

The “Enable SSH access” option is for system debugging. If you enable this option, the system will allow SSH access. The SSH connection require using the

username “admin” and the super administrator’s password. This option is turned off by default. It is recommended to turn off this option when debugging is

not required.

Tick “Enable remote SSH” option, the system will allow remote SSH access via the GDMS platform. This option is turned off by default, and it is strongly

recommended to turn off this option when remote troubleshooting is not required.

SSH Access

This option is used for system debugging. Once enabled, UCM will allow SSH access. The SSH connection requires super

Enable SSH Access administrator's username and password. The default setting is "No". It is recommended to set it to "No" if there is no need for

debugging.
 If this option is enabled, remote SSH access will be allowed through the Feedback platform. It is strongly recommended to keep this
Enable Remote SSH
disabled unless remote troubleshooting is necessary.

Access Token Please enter the token to request SSH data.

LDAP Server
The UCM630X has an embedded LDAP/LDAPS server for users to manage the corporate phonebook in a centralized manner.

By default, the LDAP server has generated the first phonebook with PBX DN “ou=pbx,dc=pbx,dc=com” based on the UCM630X user extensions

already.

Users could add new phonebook with a different Phonebook DN for other external contacts. For example, “ou=people,dc=pbx,dc=com”.

All the phonebooks in the UCM630X LDAP server have the same Base DN “dc=pbx,dc=com”.

Term Explanation:

cn= Common Name

ou= Organization Unit

dc= Domain Component

These are all parts of the LDAP Data Interchange Format, according to RFC 2849, which is how the LDAP tree is filtered.

If users have the Grandstream phone provisioned by the UCM630X, the LDAP directory will be set up on the phone and can be used right away for users to

access all phonebooks.

Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP server on the UCM630X. If the UCM630X has

multiple LDAP phonebooks created, in the LDAP client configuration, users could use “dc=pbx,dc=com” as Base DN to have access to all phonebooks on the

UCM630X LDAP server, or use a specific phonebook DN, for example “ou=people,dc=pbx,dc=com”, to access to phonebook with Phonebook DN

“ou=people,dc=pbx,dc=com ” only.

UCM can also act as an LDAP client to download phonebook entries from another LDAP server.

To access the LDAP server and client settings, go to Web GUI🡪Settings🡪LDAP Server.

LDAP Server Configurations

The following figure shows the default LDAP server configurations on the UCM630X.
 LDAP Server Configurations

The UCM630X LDAP server supports anonymous access (read-only) by default. Therefore, the LDAP client does not have to configure a username and

password to access the phonebook directory. The “Root DN” and “Root Password” here are for LDAP management and configuration where users will need

to provide for authentication purposes before modifying the LDAP information.

The default phonebook list in this LDAP server can be viewed and edited by clicking on/for the first phonebook under LDAP Phonebook.

The UCM630X support secure LDAP (LDAPS) where the communication is encrypted and secure.

Default LDAP Phonebook DN

 Default LDAP Phonebook Attributes

LDAP Phonebook

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook
from the LDAP server. The first phonebook with default phonebook dn “ou=pbx,dc=pbx,dc=com” displayed on the LDAP server page is for extensions in this

PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUI🡪Extension/Trunk🡪Extensions first.

The default LDAP phonebook will then be updated automatically.

LDAP Server🡪LDAP Phonebook

Add new phonebook

A new sibling phonebook of the default PBX phonebook can be added by clicking on “Add” under “LDAP Phonebook” section.
 Add LDAP Phonebook

Configure the “Phonebook Prefix” first. The “Phonebook DN” will be automatically filled in. For example, if configuring “Phonebook Prefix” as “people”,

the “Phonebook DN” will be filled with “ou=people,dc=pbx,dc=com”.

Once added, users can select to edit the phonebook attributes and contact list (see figure below) or select to delete the phonebook.

Edit LDAP Phonebook

Import phonebook from your computer to LDAP server

Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.

Import Phonebook

The file to be imported must be a CSV, VCF or XML file with UTF-8 encoding. Users can open the file with Notepad and save it with UTF-8 encoding.

Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required. Users could export a phonebook file from the

UCM630X LDAP phonebook section first and use it as a sample to start with.

Phonebook CSV File Format

The Phonebook DN field is the same “Phonebook Prefix” entry as when the user clicks on “Add” to create a new phonebook. Therefore, if the user enters

“phonebook” in “Phonebook DN” field in the CSV file, the actual phonebook DN “ou=phonebook,dc=pbx,dc=com” will be automatically created by the
UCM630X once the CSV file is imported.

In the CSV file, users can specify different phonebook DN fields for different contacts. If the phonebook DN already exists on the UCM630X LDAP

Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM630X LDAP Phonebook,

a new phonebook with this phonebook DN will be created.

The sample phonebook CSV file in above picture will result in the following LDAP phonebook in the UCM630X.

LDAP Phonebook After Import

As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP phonebook section, users cannot import contacts
with Phonebook DN field “pbx” if existed in the CSV file.

Export phonebook to your computer from UCM630X LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook. The exported phonebook can
be used as a record or a sample CSV, VFC or XML file for the users to add more contacts in it and import to the UCM630X again.

Export Selected LDAP Phonebook

LDAP Settings

Prerequisites to support contacts sync-up to IP Phones, UCM needs to support the following:

1. If Cloud IM is enabled, UCM can send remote UCM’s contacts to each end device.

2. Contacts from remote UCM can be synced by Cloud IM or LDAP sync via trunk. The contacts data must be complete and consistent.

3. If Cloud IM is enabled, the contacts sent from UCM to end device should integrate Cloud IM contacts.

4. If Cloud IM is disabled, the contacts sent from UCM to end device should only contain contacts on the UCM.
To support contacts sync-up to Wave, it allows Wave to obtain enterprise contacts from Cloud IM or LDAP. On UCM SIP peer trunk, if LDAP sync is

enabled, end point can obtain remote UCM extensions’ info via LDAP. Also, it will allow configuring whether to sync up LDAP contacts on Wave so that
Wave doesn’t receive duplicate contacts info.

Under UCM webUI🡪 System Settings🡪 LDAP Server, click on “LDAP Settings”, option “Wave enable LDAP phonebook” is available for configuration.
If enabled, all Wave users on this UCM will display LDAP contacts. Otherwise, it will not display.

LDAP Settings

Please note the LDAP contacts displayed on Wave will exclude the duplicate contacts from Cloud IM.

Display LDAP Contacts on Wave

LDAP Client Configurations

The configuration on LDAP client is useful when you use other LDAP servers. Here we provide an example on how to configure the LDAP client on the

UCM.

Assuming the remote server base dn is “dc=pbx,dc=com”, configure the LDAP client as follows:

LDAP Client Type

Phonebook Name Enter a name for the phonebook

Server Address The IP address of the LDAP server

Base DN Enter the base domain name.

Username Enter the username used to authenticate into the LDAP server, if authentication is required.

Password Enter the password used to authenticate into the LDAP server, if authentication is required.

Filter Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))

 Port Enter the port number. Default port is 389

LDAP Number Attributes Enter the number attributes for the remote server.

If "None" is selected, LDAP phonebooks will not automatically update. Otherwise, LDAP phonebooks will automatically
Automatic Update Cycle
update at 00:00 / 12:00 AM with the selected frequency.

LDAP Name Attributes Enter the name attributes for the remote server.

Client Type Choose the client type. For encrypted data transfer please choose LDAPS.

LDAP Client CA Cert LDAP Client Public Certification

LDAP Client Private Key LDAP Client Private Certification

The UCM can automatically update the phonebook, by configuring the ‘LDAP Automatic Update Cycle’. Available options are: 1 day/2days/7 days. It is set

to ‘None’ by default.

The following figure gives a sample configuration for UCM acting as a LDAP client.

LDAP Client Configurations

To configure Grandstream IP phones as the LDAP clients for UCM, please refer to the following example:

Server Address: The IP address or domain name of the UCM

Base DN: dc=pbx,dc=com

Username: cn=admin,dc=pbx,dc=com

Password: admin (by default)

LDAP Name Attribute: CallerIDName Email Department FirstName LastName

LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax

LDAP Number Filter: (AccountNumber=%)

LDAP Name Filter: (CallerIDName=%)

LDAP Display Name: AccountNumber CallerIDName

LDAP Version: If existed, please select LDAP Version 3

Port: 389
The following figure shows the configuration information on a Grandstream GXP2170 to successfully use the LDAP server as configured in [Figure 37:

LDAP Server Configurations].

GXP2170 LDAP Phonebook Configuration

The UCM63xx LDAP server is no longer supporting the anonymous binding of the LDAP client.

AD Client Type

Phonebook Name Enter a name for the phonebook

Server Address The IP address of the AD server

Base DN Enter the base domain name.

Username Enter the username used to authenticate into the LDAP server, if authentication is required.

Password Enter the password used to authenticate into the LDAP server, if authentication is required.

Filter Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))

Port Enter the port number. Default port is 389

 AD Attributes AccountNumber must be included if the default configuration is used.

If "None" is selected, LDAP phonebooks will not automatically update. Otherwise, LDAP phonebooks will automatically
Automatic Update Cycle
update at 00:00 / 12:00 AM with the selected frequency.

Host Name Enter the host name of the remote AD server.

Time Settings

Automatic Date and Time

The current system time on the UCM630X can be found under Web GUI🡪System Status🡪Dashboard🡪PBX Status.

To configure the UCM630X to update time automatically, go to Web GUI🡪System Settings🡪Time Settings🡪Automatic date and Time.

The configurations under Web GUI🡪Settings🡪Time Settings🡪 Automatic date and Time page require reboot to take effect. Please consider configuring auto

time updating related changes when setting up the UCM630X for the first time to avoid service interrupt after installation and deployment in production.

Remote NTP Specify the URL or IP address of the NTP server for the UCM630X to synchronize the date and time. The default NTP server is

Server pool.ntp.org.

Enable DHCP If set to “Yes”, the UCM630X can get provisioned for Time Zone from DHCP Option 2 in the local server automatically. The default
Option 2 setting is “Yes”.

Enable DHCP If set to “Yes”, the UCM630X can get provisioned for NTP Server from DHCP Option 42 in the local server automatically. This will

Option 42 override the manually configured NTP Server. The default setting is “Yes”.

Time Zone Select the proper time zone option so the UCM630X can display correct time accordingly.

Table 21: Time Auto Updating

Set Date and Time

To manually set the time on the UCM630X, go to Web GUI🡪System Settings🡪Time Settings🡪Set Date and Time. The format is YYYY-MM-DD
HH:MM:SS.

Time Auto Updating

 Note: Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would like to reboot the UCM630X and keep
the manually setup time setting, please make sure “Remote NTP Server”, “Enable DHCP Option 2” and “Enable DHCP Option 42” options under Web

GUI🡪Settings🡪Time Settings🡪Auto Time Updating page are unchecked or set to empty. Otherwise, time auto updating settings in this page will take effect after

reboot.

NTP Server

The UCM630X can be used as an NTP server for the NTP clients to synchronize their time with. To configure the UCM630X as the NTP server, set “Enable

NTP server” to “Yes” under Web GUI🡪System Settings🡪Time Settings🡪NTP Server. On the client side, point the NTP server address to the UCM630X
IP address or host name to use the UCM630X as the NTP server.

Office Time

On the UCM630x, the system administrator can define “office time” which can be used to configure time condition for extension call forwarding and inbound

rules. To configure office time, log in to the Web GUI, enter the System Settings🡪Time Settings🡪Office Time, and click the “Add” button to see the
following configuration page.

Create New Office Time

Start Time Configure the start time for office hour.

End Time Configure the end time for office hour

Week Select the workdays in one week.

Show Advanced Options Check this option to show advanced options. Once selected, please specify “Month” and “Day” below.

Month Select the months for office time.

Day Select the workdays in one month.

 Table 22: Create New Office Time

Select “Start Time”, “End Time” and the day for the “Week” for the office time. The system administrator can also define month and day of the month as

advanced options. Once done, click on “Save” and then “Apply Change” for the office time to take effect. The office time will be listed in the web page as the

figure shows below.

The office hour feature support import/export CSV files.

Settings🡪Time Settings🡪Office Time

Click on to edit the office time.

Click on to delete the office time.

Click on “Delete” to delete multiple selected office times at once.

Holiday

On UCM, the system administrator can define “holidays” which can be used to configure time condition for extension call forwarding and inbound rules. To

configure office time, log in to the Web GUI, enter the System Settings🡪Time Settings🡪Holiday, and click the “Add” button to see the following
configuration page.
 Create New Holiday

Name Specify the holiday name to identify this holiday.

Holiday Memo Create a note for the holiday.

Month Select the month for the holiday.

Select the Year for the holiday.

Year
Note: In the "Year" option, select "All" to set annual fixed holiday information.

Day Select the day for the holiday.

Show Advanced Options Check this option to show advanced options. If selected, please specify the days as holiday in one week below.

Week Select the days as holiday in one week.

Time Select the time on which the holiday starts.

Enter holiday “Name” and “Holiday Memo” for the new holiday. Then select “Month”, “Day” and the exact “Hour”. The system administrator can also define
days in one week as advanced options. Once done, click on “Save” and then “Apply Change” for the holiday to take effect. The holiday will be listed in the

web page as the figure shows.

 The Holiday feature support import/export CSV files.

Settings🡪Time Settings🡪Holiday

Click on to edit the holiday.

Click on to delete the holiday.

Click on “Delete” to delete multiple selected holidays at once.

Email Settings

Email Settings

The Email application on the UCM630X can be used to send out alert event Emails, Voicemail (Voicemail-To-Email) etc. The configuration parameters can

be accessed via Web GUI🡪System Settings🡪Email Settings🡪Email Settings.

TLS Enable Enable or disable TLS during transferring/submitting your Email to another SMTP server. The default setting is “Yes”.

Select Email type.

MTA: Mail Transfer Agent. The Email will be sent from the configured domain. When MTA is selected, there is no need to set up
Type SMTP server for it or no user login is required. However, the Emails sent from MTA might be considered as spam by the target

SMTP server.

Client: Submit Emails to the SMTP server. A SMTP server is required, and users need login with correct credentials.

Domain Specify the domain name to be used in the Email when using type “MTA”.

SMTP Server Specify the SMTP server when using type “Client”.

Enable SASL Authentication. When disabled, UCM will not try to use the username and password for mail client login authentication.
Enable SASL
Most of the mail server requires login authentication while some others private mail servers allow anonymous login which requires
Authentication
disabling this option to send Email as normal. For Exchange Server, please disable this option.

Username Username is required when using type “Client”. Normally it is the Email address.

Password Password to login for the above Username (Email address) is required when using type “Client”.

Enable Email- Monitors the inbox of the configured email address for the specified subject. If enabled, the UCM will get a copy of the attachment

to-Fax from the email and send it to the XXX extension by fax. The attachment must be in PDF/TIF/TIFF format.
 Email-to-Fax
Blacklist/Whiteli The user can enable the Email-to-Fax Blacklist or Email-to-Fax Whitelist.

st

Email-to-Fax Select the email subject format to use for emails to fax. XXX refers to the extension that the fax will be sent to. This extension can
Subject Format only contain numbers.

Internal
Email address blacklist/whitelist for local extensions.
Black/Whitelist

External

Blacklist/Whiteli Email address blacklist/whitelist for non-local contacts. Separate multiple addresses with semicolon (;) (i.e.”xxx;yyy”).

st

Fax Sending

Success/Failure If enabled, the UCM will send an email notification to the sender about the fax sending result.

Confirmation

POP/POP3 Configure the POP/POP3 server address for the configured username
Server Address Example: pop.gmail.com

POP/POP3 Configure the POP/POP3 server port for the configured username

Server Port Example: 995

Display Name Specify the display name in the FROM header in the Email.

Specify the sender’s Email address.

Sender
For example: pbx@example.mycompany.com.

Table 24: Email Settings

The following figure shows a sample Email setting on the UCM630X, assuming the Email is using 192.168.6.202 as the SMTP server.
 UCM630X Email Settings

Once the configuration is finished, click on “Test”. In the prompt, fill in a valid Email address to send a test email to verify the Email settings on the

UCM630X.

Email Templates

The Email templates on the UCM630X can be used for email notification, the configuration parameters can be accessed via Web GUI🡪Settings🡪Email

Settings🡪Email Templates.

Users can customize email templates for password reset, voicemail, meeting scheduling, extensions, fax, meeting report, PMS, CDR, emergency call, missed

calls, alert events, call queue statistics and etc.

Click on icon to edit the template.

 Email Template

Note

The “Multimedia Meeting Schedule” template is improved. Click on “Edit” for this template to view the improved default template.

Added “Edge” and “Safari” as supported browser.

Added “Download Wave” button for user to download Wave app from: https://fw.gdms.cloud/wave/download/

Improved descriptions

Email Footer Hyperlink

Under UCM Web GUI🡪System Settings🡪Email Settings🡪Email Footer Hyperlink, users could edit the text and URL to modify the email footer
hyperlink.

Email Footer Hyperlink

Email Send Log

Under UCM Web GUI🡪System Settings🡪Email Settings🡪Email Send Log, users could search, filter, and check whether the Email is sent out successfully
or not. This page will also display the corresponding error message if the Email is not sent out successfully.
 Email Send Log

Start Time Enter the start time for the filter

End Time Enter the end time for the filter

Receivers Enter the email recipient, while searching for multiple recipients, please separate them with a comma and no spaces.

Send Result Enter the status of the send result to filter with

Return Code Enter the email code to filter with

Select the email module to filter with from the drop-down list, which contains:

● All Modules

● Extension

● Voicemail
Email Send Module ● Conference Schedule

● User Password

● Alert Events

● CDR

● Test

Table 25: Email Log – Display Filter

Email logs will be shown at bottom of the “Email Send Log” page, as shown in the following figure.
 Email Logs

Below are the codes returned when sending emails and their description:

Table 26: Email Codes

Code Description

250 Mail sent successfully

Address format parsing error, 501 will be returned when there are

unacceptable characters in the recipient’s email address in MTA mode.

501
Please check if the recipient’s email address format is correct. The “sender”
configured on the client is your mail account.

The username and password verification in the client mode is incorrect.

535
Please check whether the username and password are configured correctly.

Possible reasons:

1. The recipient’s mailbox username does not exist or is in a banned state,

please check whether the email recipient is the correct email address.

2. The number of destination addresses sent by the sender exceeds the

550
maximum limit per day and is temporarily blacklisted. Please reduce the
sending frequency or try again the next day.

3. The sender’s IP does not pass the SPF permission test of the sending
domain. Emails sent in MTA mode may return this error code even if they
are sent.

552 The sent email is too large or the email attachment type is prohibited
 Code Description

The sender and the email account are inconsistent, please configure the
553
sender as your email account correctly.

The email was identified as spam. Please reduce the sending frequency or
554
try again the next day

This indicates that there is no return code.

If the sending result is “deferred”, the general reason is that the mail service

area is configured incorrectly. Please check whether the server configuration

is correct.

none
If the sending result is “bounced”, the general reason is that the receiving

email address domain name is wrong, please check whether the email

recipient is the correct email address. If it is in MTA mode, please check

whether the “domain” is configured to be in the same domain name as the

“recipient”.

SMS Settings

SMS Configuration

Configuring the SMS feature on the UCM6300 series allows the administrators to enable two-factor authentication, to send alerts, and meeting notices.

SMS Settings

Enable SMS Tick this box to enable SMS service.

SMS Carrier Choose the SMS carrier:

 ● Amazon

● Twilio

Region Choose the region.

Account ID Enter the ID of the account created at the carrier.

Secret Enter the secret code.

From Enter the number phone allocated for the UCM.

SMS Template

The template of the SMS can be modified in “SMS Template” tab. Please note that carriers may require to pre-register the templates for SMS that the UCM
will send. Refer to the Amazon and Twilio documentation for more information.

SMS Templates

SMS Delivery Log

All the SMS messages sent will be logged in the following tab.
 SMS Delivery Log

HA
Dual-system hot standby provides a highly reliable and fault-tolerant solution for enterprises using the UCM6300 series/UCM6300A series. Based on two

UCM devices of the same product model and software version, one of them is in the “Active” working state in real-time, and the other is in the “Standby”

working state. The daily data on the host server will be synchronized to the standby machine in real-time, and the standby machine always monitors the

running status of the host. When the host fails, including hardware failures and severe software failures, the standby machine will immediately take over the

business and enter the “Active” working state, and Upgrade to a host to ensure that the business is not interrupted, and the call will automatically resume.

Important Note

In order to set HA, both UCMs should have static IP addresses and their network method set to “Switch”.

HA settings

The users can configure the HA under System Settings 🡪 HA settings page.
 HA Settings

Parameter Description

High Available Enable Enables/disables the HA functionality. By default, is Disabled.

After clicking the button, the active/standby switch will be

Force switch
enforced.

The master and slave static configuration of the device, The real
HA Station Type
active/standby is decided dynamically by the active / standby.

To carry the service, the main and standby computers should be

HA Virtual IP set the same, and the intranet terminal should register and use

the IP address.

HA Peer IP Local IP address of HA peer device.

Need to specify this peer MAC address while using the UCM
HA Peer MAC Address
RemoteConnect service.

The number of the heartbeat port should be consistent with the

Heartbeat Port
peer heartbeat port.

If the timeout occurs, services will be transferred over to the

Heartbeat Timeout Period (s)
Slave UCM.

Software Fault Switch Enable Software Fault Switch

Hardware Fault Switch If issues are detected with the selected connection interfaces, the

backup UCM6510 will take over services after the master/slave

 handover. If not checked, UCM will send only a fault alarm.

If enabled, HA on UCM can be used with IPv6 while compatible

Enable IPv6
with IPv4.

Table 27: HA Settings parameters

HA Status

Once the HA is configured, the user can view its status under system settings 🡪 HA 🡪 HA Status as shown below

HA Status

HA Log

The user can view the HA log through the system settings 🡪 HA 🡪 HA log page. The HA log effectively records the execution results of past full backup
actions, as well as the historical records that triggered the active/standby switchover.

Note

The UCM63xx series suppots SNMP to be able to use 3rd party monitoring tools to monitor both UCMs in HA setup.

SNMP
UCM63xx supports SNMP in case the system administrator chooses to use third party monitoring tools. These are the options available when setting up
SNMP.
SNMP Settings

SNMP Settings

Enable Tick this box to enable SNMP.

Device Name Enter the device name.

Location Enter the location.

Contact Email Address Enter the email address used to send the SNMP alerts to.

Enable SNMP Trap Proxy Tick this box to enable a proxy for SNMP Trap.

SNMP Trap Proxy Listening

The port number on which the SNMP Trap Proxy is listening on.
Port

SNMP Community

You can also create SNMP communities and affect a certain level of access. An SNMP community is a group created to aggregate many management stations.

The community name is used to authenticate and identify these machines in the NMS (Network Management System).

Create SNMP Community

Name Enter a name for the community

Select an access level:

Access Level ● Read Only: The SNMP community will be able only to read
SNMP messages.
SNMP Traps Destinations
SNMP Traps is a very useful feature when there are many network components to manage. Instead of sending requests to all the machines in the network in

order to view their SNMP logs risking slowing down or bringing the network to a complete halt, SNMP Traps can be configured so these machines can send
unrequested messages to the manager to notify it about critical events and general failures.

Create SNMP Trap Destination

Name Enter a name of your SNMP Trap destination.

IP Address Enter the SNMP Trap destination's IP address.

Port Enter the port of the SNMP Trap destination.

Community Select the community that you want

Select the type of SNMP:

● Trapsink: Select this option if you want to send SNMP v1

traps.
Type ● Trap2sink: Select this option if you want to send SNMP v2
traps.

● Informsink: Select this option if you want to send "Inform"

notifications only.

SNMP Version 3

UCM 63xx also supports SNMP v3 in case the system administrator decides to add more security to the monitoring process. SNMP v3 is a very good solution

to monitor devices that interface directly with Internet. SNMP v3 offers more security than its predecessors by hashing the authentication information,

encrypting the SNMP messages exchanged between the managed devices and the network management system which prevent eavesdropping. Also, it

prevents any data tampering which protects the integrity of the data exchanged.
 SNMP V3 Configuration

Name Set the user's name

Select the authentication protocol:

Authentication Protocol ● MD5

● SHA

Authentication Password Set the authentication password.

Select the protocol to use to encrypt the data

● DES
Privacy Protocol ● AES-128

● AES-192

● AES-256

Privacy Password Set the privacy password.

Set the group level:

Group Level ● Read Only.

● Read/Write.

SNMP Trap Proxy

SNMP Trap Proxy

Name Enter a name for the proxy server.

 IP Address Enter the proxy server's IP address.

Port Enter the proxy server's port.

RADIUS
The UCM6300 offers Radius-based authentication for the super administrator and other administrators. This requires configuring a Radius server then

enabling Radius client on the UCM6300 which can be found under System Settings → RADIUS

Enable Radius Web

Enable or disable Radius.
Access Control

As Default Login
Enable Radius as the default login method to the web UI of the UCM
Method

Radius Auth Server

Enter the IP¨address/hostname of Radius server.
Address

Radius Auth Server Enter the port of radius server

Port Default port number is: 1812

Radius Shared Secret Enter Radius Shared Secret

Maximum Number of
Enter the number of retransmissions. The interval is 1 to 5.
Retransmission

Radius Timeout (s) The maximum seconds before a session expires if there is no response from the server. The interval is between 1 to 30 seconds.
TR-069
To configure TR-069 on Grandstream devices, set the following parameters:

Parameter Description

Enable TR-069 Toggle it on to enable TR-069. It is enabled by default

ACS URL URL for TR-069 Auto Configuration Servers (ACS), e.g., http://myacs.grandstream.com

TR-069 Username ACS username for TR-069 must be the same as in the ACS configuration.

TR-069 Password ACS password for TR-069 must be the same as in the ACS configuration.

Periodic Inform Enable Enables periodic inform. If set to Yes, the device will send inform packets to the ACS.

A periodic time when UCM630X will send inform packets to TR-069 ACS server.
Periodic Inform Interval
This option is specified in seconds.

ACS Connection Request Username The username for the ACS to connect to UCM.

ACS Connection Request Password The password for the ACS to connect to UCM.

Port for incoming connection requests.

Connection Request Port
The default value is 7547.

CPE Cert File The Cert file for UCM to connect to the ACS via SSL.

CPE Cert Key The Cert key for UCM to connect to the ACS via SSL.

PROVISIONING

Overview
Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP/HTTP/HTTPS download. All Grandstream SIP
devices support a proprietary binary format configuration file and XML format configuration file. The UCM630X provides a Plug and Play mechanism to

auto-provision the Grandstream SIP devices in a zero-configuration manner by generating an XML config file and having the phone download it within the

LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.

To provision a phone, three steps are involved, i.e., discovery, configuration, and provisioning. This section explains how Zero Config works on the

UCM630X. The settings for this feature can be accessed via Web GUI🡪Other Features🡪Zero Config.
Configuration Architecture for End Point Device
Starting from firmware version 1.0.7.10, the end point device configuration in ZeroConfig is divided into the following three layers with priority from the

lowest to the highest:

Global

This is the lowest layer. Users can configure the most basic options that could apply to all Grandstream SIP devices during provisioning via Zero config.

Model

In this layer, users can define model-specific options for the configuration template.

Device

This is the highest layer. Users can configure device-specific options for the configuration for the individual device here.

Each layer also has its own structure at different levels. Please see the figure below. The details for each layer are explained in sections [Global

Configuration], [Model configuration] and [Device Configuration].

Zero Config Configuration Architecture for End Point Device

The configuration options in the model layer and device layer have all the option in global layers already, i.e., the options in the global layer is a subset of the

options in the model layer and the device layer. If an option is set in all three layers with different values, the highest layer value will override the value in the

lower layer. For example, if the user selects English for Language setting in Global Policy and Spanish for Language setting in Default Model Template, the
language setting on the device to be provisioned will use Spanish as the model layer has higher priority than the global layer. To sum up, configurations in

the higher layer will always override the configurations for the same options/fields in the lower layer when presented at the same time.

After understanding the zero-config configuration architecture, users could configure the available options for end point devices to be provisioned by the

UCM630X by going through the three layers. This configuration architecture allows users to set up and manage the Grandstream end point devices in the

same LAN area in a centralized way.

Auto-Provisioning Settings
By default, the Zero Config feature is enabled on the UCM630X for auto-provisioning. Three methods of auto-provisioning are used.
 UCM630X Zero Config

SIP SUBSCRIBE

When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM630X discovers it and then sends a NOTIFY with the

XML config file URL in the message body. The phone will then use the path to download the config file generated in the UCM630X and take the new

configuration.

DHCP OPTION 66

Route mode needs to be set to use this feature. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER

request. The UCM630X receives it and returns DHCP OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/.

The phone will then use the path to download the config file generated in the UCM630X.

mDNS

When the phone boots up, it sends out an mDNS query to get the TFTP server address. The UCM630X will respond with its address. The phone will then

send a TFTP request to download the XML config file from the UCM630X.

To start the auto-provisioning process, under Web GUI🡪Other Features🡪Zero Config🡪Zero Config Settings, fill in the auto provision information.

Auto Provision Settings

 Enable Zero
Enable or disable the zero-config feature on the PBX. The default setting is enabled.
Config

By default, this is disabled. If disabled, when the SIP device boots up, the UCM630X will not send the SIP device the URL to
Enable
download the config file, and therefore the SIP device will not be automatically provisioned by the UCM630X.
Automatic
Configuration
Note: When disabled, SIP devices can still be provisioned by manually sending NOTIFY from the UCM630X which will include the
Assignment
XML config file URL for the SIP device to download.

Auto Assign If enabled, when the device is discovered, the PBX will automatically assign an extension within the range defined in “Zero Config

Extension Extension Segment” to the device. The default setting is disabled.

Zero Config Click on the link “Zero Config Extension Segment” to specify the extension range to be assigned if “Automatically Assign
Extension Extension” is enabled. The default range is 5000-6299. Zero Config Extension Segment range can be defined in Web GUI🡪PBX

Segment Settings🡪General Settings🡪General page🡪Extension Preference section: “Auto Provision Extensions”.

Enable Pick If enabled, the extension list will be sent out to the device after receiving the device’s request. This feature is for the GXP series

Extension phones that support selecting extensions to be provisioned via the phone’s LCD. The default setting is disabled.

Click on the link “Pick Extension Segment” to specify the extension list to be sent to the device. The default range is 4000 to 4999.
Pick Extension
Pick Extension Segment range can be defined in Web GUI 🡪 PBX Settings 🡪 General Settings 🡪 General page 🡪 Extension
Segment
Preference section: “Pick Extensions”.

Pick Extension
Specify the number of minutes to allow the phones being provisioned to pick extensions.
Period (hour)

This feature allows the UCM to provision devices in different subnets other than the UCM network.

Enter subnets IP addresses to allow devices within these subnets to be provisioned. The syntax is <IP>/<CIDR>.

Examples:
Subnet Whitelist
10.0.0.1/8

192.168.6.0/24

Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are supported.

Table 28: Auto Provision Settings

Please make sure an extension is manually assigned to the phone or “Automatically Assign Extension” is enabled during provisioning. After the configuration

on the UCM630X Web GUI, click on “Save” and “Apply Changes”. Once the phone boots up and picks up the config file from the UCM630X, it will take the

configuration right away.

Discovery
Grandstream endpoints are automatically discovered after bootup. Users could also manually discover devices by specifying the IP address or scanning the

entire LAN network. Three methods are supported to scan the devices.
 PING

ARP

SIP Message (NOTIFY)

Click on “Auto Discover” under Web GUI🡪Other Features🡪Zero Config🡪Zero Config, fill in the “Scan Method” and “Scan IP”. The IP address segment

will be automatically filled in based on the network mask detected on the UCM630X. If users need to scan the entire network segment, enter 255 (for
example, 192.168.40.255) instead of a specific IP address. Then click on “Save” to start discovering the devices within the same network. To successfully

discover the devices, “Zero Config” needs to be enabled on the UCM630X Web GUI🡪Other Features🡪Zero Config🡪Auto Provisioning Settings.

Auto Discover

The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connection Status,

Create Config, Options (Edit /Delete /Update /Reboot /Access Device Web GUI) are displayed in the list.

Figure 65: Discovered Devices

When the UCM is set to “Dual” network method, the user will be able to choose which LAN interface to use for Auto-Discovery.

Auto Discover LAN1/LAN2

Firmware
In Firmware tab, users can upload to and manage firmware for endpoints. Additionally, firmware upload size limit has been increased from 300MB to 1GB.

Upload Firmware Files to the UCM

Upload New Firmware

Enable: toggles whether the UCM will provision this firmware to endpoints if they are using the UCM as the firmware server. If not enabled, the UCM
will reject requests from endpoints for this firmware.

Model: The device model for which this firmware is intended for. Only for self-reference and has no effect on provisioning.

Firmware: The firmware version of the file being uploaded. Only for self-reference and has no effect on provisioning.

Remark: Add a comment about the uploaded firmware. Only for self-reference and has no effect on provisioning.
 Choose File to Upload: Select the firmware file to upload from the user’s PC. The file name must match the firmware file name requested by the

endpoint.

Uploading Devices List

Besides the built-in discovery method on the UCM, users could prepare a list of devices on a .CSV file and upload it by clicking on the button ”Import”, after

which a success message prompt should be displayed.

Users need to make sure that the CSV file respects the format as shown on the following figure and that the entered information is correct (valid IP address,

valid MAC address, device model, and an existing account), otherwise the UCM will reject the file and the operation will fail:

Device List – CSV file Sample

Managing Discovered Devices

Sorting: Press or to sort per MAC Address, IP Address, Version, Vendor, Model or Create Config columns from lower to higher or higher to lower,
respectively.

Filter: Select a filter to display corresponding results.

All: Display all discovered devices.

Scan Results: Display only manually discovered devices. [Discovery]

IP Address: Enter device IP and press the Search button.

MAC Address: Enter device MAC and press the Search button.

Model: Enter a model name and press the Search button. Example: GXP2130.

Extension: Enter the extension number and press the Search button.

Managing Discovered Devices

From the main menu of ZeroConfig, users can perform the following operations:

Click to access the discovery menu as shown on the [Discovery] section.

Click to add a new device to the ZeroConfig database using its MAC address.

Click to delete selected devices from the zero-config database.

Click to modify selected devices.

Click to batch update a list of devices, the UCM on this case will send SIP NOTIFY message to all selected devices to update them at
once.

Click to reboot selected devices (the selected devices, should have been provisioned with extensions since the phone will authenticate the
server which is trying to send it reboot command).

Click to clear all devices configurations.

Click to upload CSV file containing a list of devices.

Click to export CSV file containing a list of devices. This file can be imported to another UCM to quickly set it up with the original UCM’s

devices.

Click to copy configuration from one device to another. This can be useful for easily replacing devices and note that this feature works only
between devices of the same model.

All these operations will be detailed in the next sections.

Global Configuration
The global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM630X no matter what the

Grandstream device model it is. It is divided into two levels:

Global Policy

Global Templates

Global Templates configuration has higher priority to Global Policy configuration.

Global Policy

Global Policy can be accessed in Web GUI🡪Other Features🡪Zero Config🡪Global Policy page. On the top of the configuration table, users can select a

category in the “Options” dropdown list to quickly navigate to the category or they can also complete the configuration by importing/exporting. The
categories are:

Localization: configure display language, data, and time.

Phone Settings: configure the dial plan, call features, NAT, call progress tones, etc.

Contact List: configure LDAP and XML phonebook download.

Maintenance: configure upgrading, web access, Telnet/SSH access, and Syslog.

Network Settings: configure the IP address, QoS, and STUN settings.

Customization: customize LCD screen wallpaper for the supported models.

Communication Settings: configure Email and FTP settings

Select the checkbox on the left of the parameter you would like to configure to activate the dropdown list for this parameter.

Global Policy Categories

The following tables list the Global Policy configuration parameters for the SIP end device.

Language settings

Language Select the LCD display language on the SIP end device.

Date and Time

Date Format Configure the date display format on the SIP end device’s LCD.

Time Format Configure the time display in 12-hour or 24-hour format on the SIP end device’s LCD.

Enable NTP To enable the NTP service.

NTP Server Configure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server.

NTP Update Interval Configure the NTP update interval.

Time Zone Configure the time zone used on the SIP end device.

Enable Daylight Saving Time Select either to enable or disable the DST.

Table 29: Global Policy Parameters – Localization

Default Call

Settings

Configure the default dial plan rule. For syntax and examples, please refer to the user manual of the SIP devices to be provisioned for
Dial Plan
more details.

Enable Call When enabled, “Do Not Disturb”, “Call Forward” and other call features can be used via the local feature code on the phone.
Features Otherwise, the ITSP feature code will be used.

Use # as Dial Key If set to “Yes”, pressing the number key “#” will immediately dial out the input digits.
 If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based
Auto Answer by on the SIP Call-Info header sent from the server/proxy.
Call-info
The default setting is enabled.

NAT Traversal Configure if NAT traversal mechanism is activated.

User Random
If set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports.
Port

General Settings

Configure call progress tones including ring tone, dial tone, second dial tone, message waiting tone, ring back tone, call waiting tone,
busy tone, and reorder tone using the following syntax:

f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];

Call Progress

Tones
Frequencies are in Hz and cadence on and off are in 10ms).

“on” is the period (in ms) of ringing while “off” is the period of silence. Up to three cadences are supported.

Please refer to the user manual of the SIP devices to be provisioned for more details

HEADSET Key Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key. Please refer to the user manual of the SIP devices to be

Mode provisioned for more details.

Table 30: Global Policy Parameters – Phone Settings

LDAP Phonebook

Select “Manual” or “PBX” as the LDAP configuration source.

If “Manual” is selected, the LDAP configuration below will be applied to the SIP end device.
Source

If “PBX” is selected, the LDAP configuration built-in from UCM630X Web GUI🡪System Settings🡪LDAP Server will be
applied.

Address Configure the IP address or DNS name of the LDAP server.

Port Configure the LDAP server port. The default value is 389.

This is the location in the directory where the search is requested to begin. Example:

Base DN dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

Configure the bind “Username” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous
Username
binds.
 Configure the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous
Password
binds.

Number Filter Configure the filter used for number lookups. Please refer to the user manual for more details.

Name Filter Configure the filter used for name lookups. Please refer to the user manual for more details.

Version Select the protocol version for the phone to send the bind requests. The default value is 3.

Specify the “name” attributes of each record that are returned in the LDAP search result.

Example:
Name Attribute
gn

cn sn description

Specify the “number” attributes of each record that are returned in the LDAP search result.

Example:
Number Attribute
telephoneNumber

telephoneNumber Mobile

Configure the entry information to be shown on the phone’s LCD. Up to 3 fields can be displayed.

Display Name Example:

%cn %sn %telephoneNumber

Max Hits Specify the maximum number of results to be returned by the LDAP server. The valid range is 1 to 3000. The default value is 50.

Specify the interval (in seconds) for the server to process the request and the client waits for the server to return. The valid range
Search Timeout
is 0 to 180. The default value is 30.

Sort Results Specify whether the searching result is sorted or not. The default setting is No.

Incoming Calls Configure to enable LDAP number searching when receiving calls. The default setting is No.

Outgoing Calls Configure to enable LDAP number searching when making calls. The default setting is No.

Configures the display name when LDAP looks up the name for an incoming call or outgoing call.
Lookup Display

Name
It must be a subset of the LDAP Name Attributes.

XML Phonebook
 Select the source of the phonebook XML server.

Disable

Disable phonebook XML downloading.

Manual

Phonebook XML
Once selected, users need to specify downloading protocol HTTP, HTTPS, or TFTP and the server path to download the
Server
phonebook XML file. The server path could be an IP address or URL, with up to 256 characters.

Local UCM Server

Once selected, click on the Server Path field to upload the phonebook XML file. Please note after uploading the phonebook

XML file to the server, the original file name will be used as the directory name and the file will be renamed as phonebook.xml

under that directory.

Phonebook Download Configure the phonebook download interval (in minutes). If set to 0, the automatic download will be disabled. The valid range is

Interval 5 to 720.

Remove manually
edited entries on If set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically removed.
download

Table 31: Global Policy Parameters – Contact List

Upgrade and Provision

 Firmware source via ZeroConfig provisioning could be a URL for external server address, local UCM directory, or USB media
if plugged into the UCM630X. Select a source to get the firmware file:

URL

If selected to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”,
“File Prefix” and “File Postfix”.

Local UCM Server

Firmware can be uploaded to the UCM630X internal storage for firmware upgrade. If selected, click on the “Manage Storage”

Firmware Source icon next to the “Directory” option, upload the firmware file, and select a directory for the end device to retrieve the firmware

file.

Local USB Media

If selected, the USB storage device needs to be plugged into the UCM630X and the firmware file must be put under a folder

named “ZC_firmware” in the USB storage root directory.

Local SD Card Media

If selected, an SD card needs to be plugged into the UCM630X and the firmware file must be put under a folder named

“ZC_firmware” in the USB storage root directory.

Upgrade via When URL is selected as firmware source, configure upgrade via TFTP, HTTP, or HTTPS.

Server Path When URL is selected as firmware source, configure the firmware upgrading server path.

File Prefix Configure the Config Server Path.

When URL is selected as firmware source, configure the firmware file postfix. If configured, only the configuration file with
Config Server Path
the matching encrypted postfix will be downloaded and flashed into the phone.

Allow DHCP Option

If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected.
43/66
 If enabled, the end point device will automatically upgrade if a new firmware is detected. Users can select automatic upgrading
by day, by week, or by minute.

By week

Once selected, specify the day of the week to check the HTTP/TFTP server for firmware upgrades or configuration files
changes.

Automatic Upgrade
By day

Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files

changes.

By minute

Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes.

Firmware Upgrade
Specify how firmware upgrading and provisioning requests are to be sent.
Rule

Zero Config Select either to enable or disable Zero Config.

Web Access

Admin Password Configure the administrator password for admin-level login.

End-User Password Configure the end-user password for the end-user level login.

Web Access Mode Select HTTP or HTTPS as the web access protocol.

Configure the port for web access.

Web Server Port
The valid range is 1 to 65535.

RTSP Port Configure the RTSP Port.

Enable UPnP
Select either to enable or disable Enable UPnP Discovery
Discovery

Login Settings Configure the login settings.

User Login Timeout Configure User Login Timeout.

Maximum Consecutive
Configure Maximum Consecutive Failed Login Attempts.
Failed Login Attempts

Login Error Lock

Configure Login Error Lock Time.
Time
Security

Enable Telnet/SSH access for the SIP end device. If the SIP end device supports Telnet access, this option controls the Telnet
Disable Telnet/SSH
access of the device; if the SIP end device supports SSH access, this option controls the SSH access of the device.

Syslog

Syslog Server Configure the URL/IP address for the Syslog server.

Syslog Level Select the level of logging for Syslog.

Send SIP Log Configure whether the SIP log will be included in the Syslog message.

Table 32: Global Policy Parameters – Maintenance

Basic Settings

Configure how the SIP end device shall obtain the IP address. DHCP or PPPoE can be selected.

DHCP

Once selected, users can specify the Host Name (option 12) of the SIP end device as DHCP client, and Vendor Class ID (option
IP Address
60) used by the client and server to exchange vendor class ID information.

PPPoE

Once selected, users need to specify the Account ID, Password, and Service Name for PPPoE.

Host Name Specify the name of the client. This field is optional but may be required by Internet Service Providers.

Vendor Class ID Used by clients and servers to exchange vendor class ID.

Account ID Enter the PPPoE account ID.

Password Enter the PPPoE Password.

Service Name Enter the PPPoE Service Name.

Advanced Setting

Layer 3 QoS Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv, or MPLS. The valid range is 0-63.

Assign the priority value of the Layer 3 QoS for RTP packets.
Layer 3 QoS For

RTP
The valid range is 0 -63.
 Assign the priority value of the Layer 3 QoS for SIP packets.
Layer 3 QoS For

SIP
The valid range is 0 -63.

Assign the VLAN Tag of the Layer 2 QoS packets.

Layer 2 QoS Tag
The valid range is 0 -4095.

Assign the priority value of the Layer 2 QoS packets.

Layer 2 QoS

Priority Value
The valid range is 0-7.

STUN Server Configure the IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Keep-Alive Select either to enable or disable Keep Alive.

Specify how often the phone will send a blank UDP packet to the SIP server to keep the “ping hole” on the NAT router open. The
Keep Alive Interval
valid range is 10-160.

Register Expiration Specify the Register Expiration.

Local SIP Port Configure Local SIP Port.

Local RTP Port Configure Local RTP Port.

Auto On-Hook
Configure Auto On-Hook Timer(s).
Timer(s)

Ring Timeout Configure Ring Timeout.

SIP Transport Select either UDP, TCP, or TLS/TCP as SIP transport protocol.

Direct IP Call Select either to disable or enable Direct IP Call support.

SIP Proxy
Select either to disable or enable SIP Proxy Compatibility Mode.
Compatibility Mode

Unregister On
Select either to disable or enable Unregister On Reboot.
Reboot

Whitelist

Whitelist Select either to enable or disable Whitelist

SIP Phone Number

Configure the SIP Phone Number Whitelist.
Whitelist

Table 33: Global Policy Parameters – Network Settings

 Wallpaper

Check this option if the SIP end device shall use 1024 x 600 resolution for the LCD screen wallpaper.

Source

Screen Resolution Configure the location where wallpapers are stored.

1024 x 600
File

If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click

to upload wallpaper file to the UCM630X.

Check this option if the SIP end device shall use 800 x 400 resolution for the LCD screen wallpaper.

Source

Screen Resolution 800 Configure the location where wallpapers are stored.

x 400
File

If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click
to upload wallpaper file to the UCM630X.

Check this option if the SIP end device shall use 480 x 272 resolution for the LCD screen wallpaper.

Source

Screen Resolution 480 Configure the location where wallpapers are stored.

x 272
File

If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click

to upload wallpaper file to the UCM630X.

Check this option if the SIP end device supports 320 x 240 resolution for the LCD screen wallpaper.

Source

Screen Resolution 320 Configure the location where wallpapers are stored.

x 240
File

If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click

to upload wallpaper file to the UCM630X.

Table 35: Global Policy Parameters – Communication Settings

Email Settings
 Check this option to configure the email settings that will be sent to the provisioned phones:

Server

The IP address of the SMTP server

Port

SMTP server port

From E-Mail address

Email address

Sender Username

SMTP Settings Username of the sender

Password Recovery Email

Email where the recovered password will be sent

Alarm receive Email 1

Email address where alarms notifications will be sent

Alarm receive Email 1

Email address where alarms notifications will be sent

Enable SSL

Enable SSL protocol for SMTP

FTP
 Check this option to configure the FTP settings that will be sent to the provisioned phones:

Storage Server Type

Either FTP or Central Storage

Server

FTP server address

FTP Port

FTP port to be used

Username

FTP username

Path

FTP Directory path

Table 34: Global Policy Parameters – Customization

Global Templates

Global Templates can be accessed in Web GUI🡪Other Features🡪Zero Config🡪Global Templates. Users can create multiple global templates with
different sets of configurations and save the templates, or click on the “Import/Export” button to add multiple global templates. Later on, when the user

configures the device in the Edit Device dialog🡪Advanced Settings, the user can select to use one of the global templates for the device. Please refer to
section [Manage Devices] for more details on using the global templates.

When creating a global template, users can select the categories and the parameters under each category to be used in the template. The global policy and the

selected global template will both take effect when generating the config file. However, the selected global template has higher priority to the global policy
when it comes to the same setting option/field. If the same option/field has a different value configured in the global policy and the selected global template,

the value for this option/field in the selected global template will override the value in global policy.

Click on “Add” to add a global template. Users will see the following configurations.

Template Name Create a name to identify this global template.

Description Describe the global template. This is optional.

Active Check this option to enable the global template.

Table 36: Create New Template

Click on to edit the global template.

The window for editing the global template is shown in the following figure. In the “Options” field, after entering the option name keyword, the options

containing the keyword will be listed. Users could then select the options to be modified under the global template.
 Edit Global Template

The added options will show in the list. Users can then enter or select the value for each option to be used in the global template. On the left side of each

added option, users can click to delete this option from the template. On the right side of each option, users can click on to reset the option value to the

default value.

Click on “Save” to save this global template.

The created global templates will show in the Web GUI🡪Other Features🡪Zero Config🡪Global Templates page. Users can click on to delete the
global template or delete multiple selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected templates.

Model configuration

Model templates

Model layer configuration allows users to apply model-specific configurations to different devices. Users could create/edit/delete a model template by
accessing Web GUI, page Other Features🡪Zero Config🡪Model Templates. If multiple model templates are created and enabled, when the user configures

the device in the Edit Device dialog🡪Advanced Settings, the user can select to use one of the model templates for the device. Please refer to section [Manage
Devices] for more details on using the model template.

For each created model template, users can assign it as the default model template. If assigned as the default model template, the values in this model template

will be applied to all the devices of this model. There is always only one default model template that can be assigned at one time on the UCM630X.

The selected model template and the default model template will both take effect when generating the config file for the device. However, the model template

has a higher priority than the default model template when it comes to the same setting option/field. If the same option/field has different value configured in

the default model template and the selected model template, the value for this option/field in the selected model template will override the value in default

model template.
 Click on “Add” to add a model template.

Model Select a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.

Template Name Create a name for the model template.

Description Enter a description for the model template. This is optional.

Default Model Select to assign this model template as the default model template. The value of the option in the default model template will be

Template overridden if another selected model template has a different value for the same option.

Active Check this option to enable the model template.

Table 37: Create New Model Template

Click on to edit the model template.

The editing window for a model template is shown in the following figure. In the “Options” field, enter the option name key word, the option that contains the

keyword will be listed. The user could then select the option to be modified under the model template.

Once added, the option will be shown in the list below. On the left side of each option, users can click on to remove this option from the model template.
On the right side of each option, users can click on to reset the option to the default value.

The user could also click on “Add New Field” to add a P-value number and the value to the configuration. The following figure shows setting P-value

“P1362” to “en”, which means the display language on the LCD is set to English. For P-value information of different models, please refer to the

configuration template here http://www.grandstream.com/support/tools

Edit Model Template

 Click on Save when done. The model template will be displayed on Web GUI🡪Other Features🡪Zero Config🡪Model Templates page.

Click on to delete the model template or click on “Delete Selected Templates” to delete multiple selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected model templates.

Click on the “Import/Export” button to upload/export the model template list in .CSV format.

To make it easier for the administrator to search through the templates, a filter button has been added the user interface. Please see the screenshot below:

Filter Model Templates

Model Update

UCM630X Zero Config feature supports provisioning all models of Grandstream SIP end devices including OEM device models.

OEM Models

Users can associate OEM device models with their original Grandstream-branded models, allowing these OEM devices to be provisioned appropriately.

Click on button.

In the Source Model field, select the Grandstream device that the OEM model is based on from the dropdown list.

For the Destination Model and Destination Vendor field, enter the custom OEM model name and vendor name.

The newly added OEM model should now be selectable as an option in Model fields.

OEM Models

To make it easier for the users to search for the model templates to download or update, a filter button has been added to the user interface.
 Filter Endpoint Models

Model Template Package List

Templates for most of the Grandstream models are built-in with the UCM630X already. Templates for Wave and Grandstream surveillance products require

users to download and install under Web GUI🡪Other Features🡪Zero Config🡪Model Update first before they are available in the UCM630X for

selection. After downloading and installing the model template to the UCM630X, it will show in the dropdown list for “Model” selection when editing the
model template.

Click on to download the template.

Click on to upgrade the model template. Users will see this icon available if the device model has a template updated in the UCM630X.

Template Management

Upload Model Template Package

In case the UCM630X is placed in the private network and Internet access is restricted, users will not be able to get packages by downloading and installing

from the remote server. Model template packages can be manually uploaded from a local device through Web GUI. Please contact Grandstream customer

support if the model package is needed for manual uploading.

Upload Model Template Manually

Device Configuration
On Web GUI, page Other Features🡪Zero Config🡪Zero Config, users could create a new device, delete existing device(s), make special configuration for a
single device, or send NOTIFY to an existing device(s).

Create New Device

Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is
discovered by the UCM630X. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to

set up each device beforehand.

Click on “Add” and the following dialog will show. Follow the steps below to create the configurations for the new device.

1. Firstly, select a model for the device to be created and enter its MAC address, IP address, and firmware version (optional) in the corresponding field.

2. Basic settings will show a list of settings based on the model selected in step 1. Users could assign extensions to accounts, assign functions to Line Keys

and Multiple-Purposed Keys if supported on the selected model.

3. Click on “save” to save the configuration for this device.

Create New Device

Manage Devices

The device manually created or discovered from Auto Discover will be listed in the Web GUI🡪Other Features🡪Zero Config🡪Zero Config page. Users
can see the devices with their MAC address, IP address, vendor, model, etc.

Manage Devices
 Click on to access the Web GUI of the phone.

Click on to edit the device configuration.

A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings. “Basic” settings have the same configurations as

displayed when manually creating a new device, i.e., account, line key, and MPK settings; “Advanced” settings allow users to configure more details in a five-

level structure.

Edit Device

A preview of the “Advanced” settings is shown in the above figure. There are five levels configurations as described in (1) (2) (3) (4) (5) below, with priority

from the lowest to the highest. The configurations in all levels will take effect for the device. If the same options are existing in different level configurations

with different values configured, the higher-level configuration will override the lower-level configuration.

1. Global Policy

This is the lowest level configuration. The global policy configured in Web GUI🡪Other Features🡪Zero Config🡪Global Policy will be applied here.
Clicking on “Modify Global Policy” to redirect to page Other Features🡪Zero Config🡪Global Policy.

2. Global Templates

Select a global template to be used for the device and click on to add. Multiple global templates can be selected, and users can arrange the priority by

adjusting orders via and . All the selected global templates will take effect. If the same option exists on multiple selected global templates, the value
in the template with higher priority will override the one in the template with lower priority. Click on to remove the global template from the selected

list.

3. Default Model Template

Default Model Template will be applied to the devices of this model. Default model template can be configured in the model template under Web

GUI🡪Other Features🡪Zero Config🡪Model Templates page. Please see the default model template option in [Table 37: Create New Model Template].

4. Model Templates

Select a model template to be used for the device and click on to add. Multiple model templates can be selected, and users can arrange the priority by

adjusting orders via and . All the selected model templates will take effect. If the same option exists on multiple selected model templates, the value

in the template with higher priority will override the one in the template with lower priority. Click on to remove the model template from the selected

list.

5. Customize Device Settings

This is the highest-level configuration for the device. Click on “Modify Customize Device Settings” and the following dialog will show.

Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options that are not in the pre-defined list, click on

“Add New Field” to add a P-value number and the value to the configuration. The above figure shows setting P-value “P1362” to “en”, which means the

display language on the LCD is set to English. The warning information on the right tells that the option matching the P-value number exists and clicking on it

will lead to the matching option. For P-value information of different models, please refer to the configuration template here

http://www.grandstream.com/sites/default/files/Resources/config-template.zip.

Select multiple devices that need to be modified and then click on ”Update Config” to batch modify devices.

If selected devices are of the same model, the configuration dialog is like the following figure. Configurations in five levels are all available for users to

modify.
 Modify Selected Devices – Same Model

If selected devices are of different models, the configuration dialog is like the following figure. Click on to view more devices of other models. Users are

only allowed to make modifications to the Global Templates and Global Policy level.

Modify Selected Devices – Different Models

Performing batch operation will override all the existing device configuration on the page.

After the above configurations, save the changes and go back to Web GUI🡪Other Features🡪Zero Config🡪Zero Config page. Users could then click on

to send NOTIFY to the SIP end point device and trigger the provisioning process. The device will start downloading the generated configuration file from
the URL contained in the NOTIFY message.
 Device List in Zero Config

On this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices.

Sample Application
Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3275 phone used by customer support

supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provision the

office phones for the customer support team.

1. Go to Web GUI🡪Other Features🡪Zero Config🡪Zero Config Settings, select “Enable Zero Config”.

2. Go to Web GUI🡪Other Features🡪Zero Config🡪Global Policy, configure Date Format, Time Format, and Firmware Source as follows.
 Zero Config Sample – Global Policy

3. Go to Web GUI🡪Other Features🡪Zero Config🡪Model Templates, create a new model template “English Support Template” for GXP2170. Add
option “Language” and set it to “English”. Then select the option “Default Model Template” to make it the default model template.

4. Go to Web GUI🡪Other Features🡪Zero Config🡪Model Templates, create another model template “Spanish Support Template” for GXP2170. Add

option “Language” and set it to “Español”.

5. After 9 devices are powered up and connected to the LAN network, use the “Auto Discover” function or “Create New Device” function to add the

devices to the device list on Web GUI🡪Other Features🡪Zero Config🡪Zero Config.

6. On Web GUI🡪Other Features🡪Zero Config🡪Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed
on the list. Click on to edit the device settings.

7. For each of the 5 phones used by English-speaking customer support, in “Basic settings” select an available extension for account 1 and click on “Save”.

Then click on the “Advanced settings” tab to bring up the following dialog. Users will see the English support template is applied since this is the default

model template. A preview of the device settings will be listed on the right side.
 Zero Config Sample – Device Preview 1

8. For the 3 phones used by Spanish support, in “Basic settings” select an available extension for account 1 and click on “Save”. Then click on the

“Advanced settings” tab to bring up the following dialog.

Zero Config Sample – Device Preview 2

Select “Spanish Support Template” in “Model Template”. The preview of the device settings is displayed on the right side and we can see the language is set
to “Español” since Model Template has the higher priority for the option “Language”, which overrides the value configured in the default model template.

9. For the GXV3275 used by the customer support supervisor, select an available extension for account 1 on “Basic settings” and click on “Save”. Users can

see the preview of the device configuration in “Advanced settings”. There is no model template configured for GXV3275.

Zero Config Sample – Device Preview 3

10. Click on “Apply Changes” to apply saved changes.

 11. On the Web GUI🡪Other Features🡪Zero Config🡪Zero Config page, click on to send NOTIFY to trigger the device to download the config file
from UCM630X.

Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM630X. 3 of the phones will be provisioned to display

Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default

language on the LCD display since it is not specified in the global policy.

EXTENSIONS

Create New User

Create New SIP Extension

To manually create a new SIP user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new window will show for users to fill in the
extension information.

Create New Extension

Extension options are divided into five categories:

Basic Settings

Media

Features

Specific Time

Wave

Follow me

Select first which type of extension: SIP Extension, IAX Extension, or FXS Extension. The configuration parameters are as follows.

General
Extension The extension number associated with the user.

Configure the CallerID Number that would be applied for outbound calls from this user.

CallerID Number Note:

The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.

Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest

level to the highest level. The default setting is "Internal".

Privilege
Note:

Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls using this rule.

Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password
SIP/IAX Password
for security purposes.

Auth ID Configure the authentication ID for the user. If not configured, the extension number will be used for authentication.

Configure Voicemail. There are three valid options, and the default option is "Enable Local Voicemail".

● Disable Voicemail: Disable Voicemail.

Voicemail ● Enable Local Voicemail: Enable voicemail for the user.

● Enable Remote Voicemail: Forward the notify message from the remote voicemail system for the user, and the local voicemail will
be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil).

Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically
Voicemail Password
generated. It is recommended to use the randomly generated password for security purposes.

Skip Voicemail
When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access.
Password
By default, this option is disabled.
Verification

Send Voicemail
Configures whether to send emails to the extension's email address to notify of a new voicemail.
Email Notification

Attach Voicemail to Configures whether to attach a voicemail audio file to the voicemail notification emails.

Email Note: When set to “Default”, the global settings in Call Features 🡪 Voicemail 🡪 Voicemail Email Settings will be used.

Keep Voicemail Whether to keep the local voicemail recording after sending them. If set to “Default”, the global settings will be used.

after Emailing Note: When set to “Default”, the global settings in Call Features 🡪 Voicemail 🡪 Voicemail Email Settings will be used.

Enable Keep-alive If enabled, an empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "No".

Keep-alive
Configure the Keep-alive interval (in seconds) to check if the host is up. The default setting is 60 seconds.
Frequency

If enabled, (1) Call Forward, Call Waiting, and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1,
Enable SCA
and (3) Private numbers can be added in Call Features🡪SCA page.
Emergency CID
CallerID name that will be used for emergency calls and callbacks.
Name

Disable This If selected, this extension will be disabled on the UCM630X.

Extension Note: The disabled extension still exists on the PBX but cannot be used on the end device.

If enabled, this extension number will be displayed in the Wave contact, otherwise, it will not be displayed, and it cannot be found in the
Sync Contact
chat, but the user can still dial this number.

User Settings

First Name Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name Configure the last name of the user. The last name can contain characters, letters, digits, and _.

Email Address Fill in the Email address for the user. Voicemail will be sent to this Email address.

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the
User Password
randomly generated password for security purposes.

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt

language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the currently available
Language
voice prompt languages on the UCM630X. To add more languages to the list, please download the voice prompt package by selecting

"Check Prompt List" under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Concurrent
The maximum endpoints which can be registered into this extension. For security concerns, the default value is 1.
Registrations

Configure the phone number for the extension, user can type the related star code for the phone number followed by the extension
Mobile Phone
number to directly call this number.
Number
For example, the user can type *881000 to call the mobile number associated with extension 1000.

Configure the user's department. The department can be configured in User Management->Address Book Management->Department

Department Management.

Job Title: The user's department position.

Contact Privileges

Same as

Department When enabled, The extension will inherit the same privilege attributed to the department it belongs to.

Contact Privileges

Contact View
Select the privileges regarding the contact view in SIP endpoints and Wave.
Privileges

SIP Extension Configuration Parameters🡪Basic Settings

SIP Settings
 Use NAT when the UCM630X is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there

NAT is a one-way audio issue, usually it is related to NAT configuration or the Firewall's support of SIP and RTP ports. The default

setting is enabled.

By default, the UCM630X will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to

Enable Direct Media negotiate with the endpoints to route the media stream directly. It is not always possible for the UCM630X to negotiate endpoint-to-

endpoint media routing. The default setting is "No".

Select DTMF mode for the user to send DTMF. The default setting is "RFC4733". If "Info" is selected, the SIP INFO message will

DTMF Mode be used. If "Inband" is selected, a-law or u-law are required. When "Auto" is selected, RFC4733 will be used if offered, otherwise

"Inband" will be used.

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. The “User=Phone” parameter will

TEL URI be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel” will be

used instead of “SIP” in the SIP request.

Alert-Info When present in an INVITE request, the alert-Info header field specifies an alternative ring tone to the UAS.

Enable T.38 UDPTL Enable or disable T.38 UDPTL support.

SRTP Enable SRTP for the call. The default setting is disabled.

Select the jitter buffer method.

● Disable: Jitter buffer will not be used.

Jitter Buffer ● Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size")

● Adaptive: Jitter buffer with an adaptive size (no more than the value of "max jitter buffer").

● NetEQ: Dynamic jitter buffer via NetEQ.

Configure to enable Packet Loss Retransmission.

Packet Loss ● NACK

Retransmission ● NACK+RTX(SSRC-GROUP)

● OFF

Video FEC Check to enable Forward Error Correction (FEC) for Video.

Audio FEC Check to enable Forward Error Correction (FEC) for Audio.

If enabled, the UCM will send CN packets for silence suppression after a successful CN negotiation in the SIP SDP. If the client
Silence Suppression
endpoint's OPUS codec supports the reception of DTX packets, the UCM will send DTX packets instead.

FECC Configure to enable Remote Camera Management.

ACL Policy Access Control List manages the IP addresses that can register to this extension.

● Allow All: Any IP address can register to this extension.

 ● Local Network: Only IP addresses in the configured network segments can register to this extension. Press “Add Local
Network Address” to add more IP segments.

The following encryption protocols can be used to encrypt an RTP stream.

● AES_CM_128_HMAC_SHA1_80 (This is the default used protocol)

SRTP Crypto Suite ● AES_256_CM_HMAC_SHA1_80

● AEAD_AES_128_GCM

● AEAD_AES_256_GCM

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722,
Codec Preference
G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX and VP8.

SIP Extension Configuration Parameters🡪Media

Call Transfer

Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are
Presence Status
possible: “Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”. More details at [PRESENCE].

Internal Calls & External Calls

Enable and configure the Call Forward Unconditional target number. Available options for target number are:

● “None”: Call forward deactivated.

● “Extension”: Select an extension from the dropdown list as CFU target.

● “Custom Number”: Enter a customer number as a target. For example: *97.

● “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected
Call Forward
extension.
Unconditional
● “Ring Group”: Select a ring group from the dropdown list as CFU target.

● “Queues”: Select a queue from the dropdown list as CFU target.

● “Voicemail Group”: Select a voicemail group from the dropdown list as CFU target.

● Custom Prompt: The call will be forwarded to a custom prompt.

The default setting is “None”.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time

conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time

and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:
CFU Time Condition

● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward No Configure the Call Forward No Answer target number. Available options for target number are:

Answer
● “None”: Call forward deactivated.

● “Extension”: Select an extension from the dropdown list as CFN target.

● “Custom Number”: Enter a customer number as a target. For example: *97.

 ● “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected
extension.

● “Ring Group”: Select a ring group from the dropdown list as CFN target.

● “Queues”: Select a queue from the dropdown list as CFN target.

● “Voicemail Group”: Select a voicemail group from the dropdown list as CFN target.

● Custom Prompt: The call will be forwarded to a custom prompt.

The default setting is “None”.

Select time condition for Call Forward No Answer. The available time conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’,

‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific

Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:
CFN Time Condition

● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Configure the Call Forward Busy target number. Available options for target number are:

● “None”: Call forward deactivated.

● “Extension”: Select an extension from the dropdown list as CFB target.

● “Custom Number”: Enter a customer number as a target. For example: *97

● “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected
Call Forward Busy extension.

● “Ring Group”: Select a ring group from the dropdown list as CFB target.

● “Queues”: Select a queue from the dropdown list as CFB target.

● “Voicemail Group”: Select a voicemail group from dropdown list as CFB target.

● Custom Prompt:

The default setting is “None”.

Select time condition for Call Forward Busy. The available time conditions ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’,

‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out

of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:
CFB Time Condition

● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Do Not Disturb If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.

Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
DND Time Condition
● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
 If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by

new lines. Pattern matching is supported.

DND Whitelist ● Z match any digit from 1-9.

● N match any digit from 2-9.

● X match any digit from 0-9.

Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

● Z match any digit from 1-9.

FWD Whitelist
● N match any digit from 2-9.

● X match any digit from 0-9.

CC Settings

If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party
Enable CC
failed for some reason. By default, it is disabled.

Two modes for Call Completion are supported:

● Normal: This extension is used as an ordinary extension.

CC Mode
● For Trunk: This extension is registered from a PBX.

The default setting is “Normal”.

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the

CC Max Agents maximum number of CC requests this channel can make.

The minimum value is 1.

Configure the maximum number of monitor structures that may be created for this device. In other words, this number tells how

CC Max Monitors many callers may request CC services for a specific device at one time.

The minimum value is 1.

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound,
Ring Simultaneously
the register number will be used to be displayed for the external number as the caller ID number.

Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.
External Number
This field accepts only letters, numbers, and special characters + = * #.

Time Condition for

Ring the external number simultaneously along with the extension based on this time condition.
Ring Simultaneously

Use callee DOD on

Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.
FWD or RS

Monitor privilege control

Call Montoring Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension

Whitelist using feature code.

Allow Operator Panel

Configure whether this extension can be monitored by the Operator Panel administrator.
Monitoring

Seamless transfer privilege control

Allowed to seamless Any extensions on the UCM can perform a seamless transfer. When using the Pickup Incall feature, only extensions available on the

transfer “Selected Extensions” list can perform a seamless transfer to the edited extension.

PMS Remote Wakeup Whitelist

Select the extensions

that can set wakeup

Selected extensions can set a PMS wakeup service for this extension via feature code.
service for other

extensions

Other Settings

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up

(voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X. The valid range is between 5
Ring Timeout
seconds and 600 seconds.

Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under
Auto Record
Web GUI🡪CDR🡪Recording Files.

● If set to “yes”, users can skip entering the password when making outbound calls.
Skip Trunk Auth ● If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

● If set to “No”, users will be asked to enter the password when making outbound calls.

Time Condition for If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering the password when making

Skip Trunk Auth outbound calls.

Dial Trunk Password Configure personal password when making outbound calls via the trunk.

Check to enable Hot-Desking Mode on the extension. Hot-Desking allows using the same endpoint device and logs in using

Support Hot-Desking extension/password combination. This feature is used in scenarios where different users need to use the same endpoint device during

Mode a different time of the day for instance. If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will be

changed to the same as Extension.

If enabled, the extension will be added to the LDAP Phonebook PBX list.
Enable LDAP
Default is enabled.

Use MOH as IVR If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of the regular

ringback tone ringback tone.

 Music On Hold Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.

Call Duration Limit Check to enable and set the call limit the duration.

Maximum Call
The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
Duration (s)

The Maximum The maximum number of simultaneous calls that the extension can have.

Number of Call Lines 0 indicates no limit.

Outgoing Call If enabled, if the number of outbound calls exceed the configured threshold within the specified period, further outbound calls will be

Frequency Limit not be allowed.

Send PCPID Header If enabled, this extension's SIP INVITE messages will contain the P-Called-Party-ID (PCPID) header if the callee is a SIP device.

Period (m) The period of outgoing call frequency limit. The valid range is from 1 to 120. The default value is 1.

Max Number of Calls Set the maximum number of outgoing calls in a period. The valide tange is from 1 to 20. The default value is 5.

Enable Auto-Answer
If enabled, the extension will support auto-answer when indicated by Call-info/Alert-info headers.
Support

Allows calls to the extension even when it is already in a call. This only works if the caller is directly dialing the extension. If
Call Waiting
disabled, the CC service will take effect only for unanswered and timeout calls.

If enabled, when the extension has concurrent registrations on multiple devices, upon incoming call or meeting invite ringing, if one
Stop Ringing
end device rejects the call, the rest of the devices will also stop ringing. By default, it’s disabled.

Email Missed Call

If enabled, the log of missed calls will be sent to the extension’s configured email address.
Log

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

● Default: All missed calls will be sent in email notifications.

Missed Call Type
● Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

● Missed External Call: Only missed calls from trunks will be sent in email notifications.

SIP Extension Configuration Parameters🡪Features

Specific Time

Time Condition Click to add Time Condition to configure a specific time for this extension.

Table 41: SIP Extension Configuration Parameters🡪Specific Time

Normal
 Enable Wave Enable Wave for the specific extension.

Wave Welcome Email Wave Welcome Email template.

Wave

Download Link https://fw.gdms.cloud/wave/download/

SIP Extension Configuration Parameters🡪Wave

Important Note

Please note that the access to the QR code using the UCM6300 web UI using the admin account has been removed for security and confidentiality reasons.

Follow Me

Enable Configure to enable or disable Follow Me for this user.

If the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of
Skip Trunk Auth
the Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering If enabled, call will need to be confirmed after answering.

Enable Destination Configure to enable destination.

Default Destination The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow Me Use the callee DOD number as CID if configured Follow Me numbers are external numbers.

Play Follow Me Prompt If enabled, the Follow Me prompt tone will be played.

Add a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan
New Follow Me Number
should have permissions to dial the defined external number.

Dialing Order This is the order in which the Follow Me destinations will be dialed to reach the user.

Create New IAX Extension

The UCM630X supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony sessions between servers and terminal devices.

IAX is like SIP but also has its own characteristic. For more information, please refer to RFC 5465.
To manually create a new IAX user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new dialog window will show for users which

need to make sure first to select the extension type to be IAX Extension before proceeding to fill in the extension information. The configuration parameters
are as follows.

General

Extension The extension number associated with the user.

Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your
CallerID Number
outbound Caller ID may be limited by your VoIP provider.

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the

lowest level to the highest level. The default setting is “Internal”.

Privilege
Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls using this

rule.

Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this
SIP/IAX Password
password for security purposes.

Configure Voicemail.

There are three valid options, and the default option is “Enable Local Voicemail”.
Voicemail
● Disable Voicemail: Disable Voicemail.

● Enable Local Voicemail: Enable voicemail for the user.

Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is
Voicemail Password
automatically generated. It is recommended to use the randomly generated password for security purposes.

Skip Voicemail Password When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail

Verification access. By default, this option is disabled.

Send Voicemail Email

Configures whether to send emails to the extension’s email address to notify of a new voicemail.
Notification

Attach Voicemail to Email Configures whether to attach a voicemail audio file to the voicemail notification emails.

Keep Voicemail after

Only applies if extension-level or global Send Voicemail to Email is enabled.
Emailing

If selected, this extension will be disabled on the UCM630X.

Disable This Extension
Note: The disabled extension still exists on the PBX but cannot be used on the end device.

User Settings

First Name Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name Configure the last name of the user. The last name can contain characters, letters, digits, and _.
Email Address Fill in the Email address for the user. Voicemail will be sent to this Email address.

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to
User Password
use the randomly generated password for security purposes.

Select the voice prompt language to be used for this extension. The default setting is “Default” which is the selected voice

prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the
Language
currently available voice prompt languages on the UCM630X. To add more languages to the list, please download the voice

prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Mobile Phone Number Configure the Mobile number of the user.

IAX Settings

Max Number of Calls Configure the maximum number of calls allowed for each remote IP address.

Configure to enable/disable requiring call token. If set to “Auto”, it might lock out users who depend on backward
Require Call Token
compatibility when peer authentication credentials are shared between physical endpoints. The default setting is “Yes”.

SRTP Enable SRTP for the call. The default setting is disabled.

Access Control List manages the IP addresses that can register to this extension.

ACL Policy ● Allow All: Any IP address can register to this extension.

● Local Network: Only IP addresses in the configured network segments can register to this extension.

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726,
Codec Preference
G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX and VP8.

Call Transfer

Configure the Call Forward Unconditional target number. If not configured, the Call Forward Unconditional feature is
Call Forward Unconditional
deactivated. The default setting is deactivated.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The

available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or

Holiday”, and “Specific”.

Note:
CFU Time Condition
● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time
Condition for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is
Call Forward No Answer
deactivated. The default setting is deactivated.
 Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”,

“Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

CFN Time Condition ● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time
Condition for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Configure the Call Forward Busy target number. If not configured, the Call Forward Busy feature is deactivated. The
Call Forward Busy
default setting is deactivated.

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”,

“Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.
CFB Time Condition
● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time
Condition for a specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Do Not Disturb If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.

DND Time Condition The time condition of DND. The DND will take effect while the time condition is satisfied.

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be

separated by new lines. Pattern matching is supported.

DND Whitelist ● Z match any digit from 1-9.

● N match any digit from 2-9.

● X match any digit from 0-9.

Calls from users in the forward whitelist will not be forwarded.

Pattern matching is supported.

FWD Whitelist ● Z match any digit from 1-9.

● N match any digit from 2-9.

● X match any digit from 0-9.

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for
Ring Simultaneously
outbound, the register number will be used to be displayed for the external number as the caller ID number.

External Number Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.

Time Condition for Ring Ring the external number simultaneously along with the extension based on this time condition.
Simultaneously

Use callee DOD on FWD or RS Use the callee’s DOD number as CallerID on Outgoing Forwarding or Ring Simultaneously calls.

Monitor Privilege Control

Call Monitoring Whitelist Members of the list can spy on this extension via feature codes.

Seamless transfer privilege control

Allowed to seamless transfer Members of the list can seamlessly transfer via feature code.

Other Settings

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang
up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be

configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General
Ring Timeout
Preference. The valid range is between 5 seconds and 600 seconds.

Note: If the endpoint also has a ring timeout configured, the actual ring timeout used is the shortest time set by either

device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be
Auto Record
accessed under Web GUI🡪CDR🡪Recording Files.

● If set to “Yes”, users can skip entering the password when making outbound calls.

● If set to “By Time”, users can skip entering the password when making outbound calls during the selected time
Skip Trunk Auth
condition.

● If set to “No”, users will be asked to enter the password when making outbound calls.

Time Condition for Skip Trunk If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can skip entering the password when

Auth making outbound calls.

Dial Trunk Password Configure personal password when making outbound calls via the trunk.

Enable LDAP If enabled, the extension will be added to LDAP Phonebook PBX lists.

Music On Hold Configure the Music On Hold class to suggest to the bridged channel when putting them on hold.

If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of
Use MOH as IVR ringback tone
the regular ringback tone.

Call Duration Limit Check to enable and set the call limit the duration.

Maximum Call Duration (s) The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

Email Missed Calls Send a log of missed calls to the extension’s email address.
 If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

● Default: All missed calls will be sent in email notifications.

Missed Call Type
● Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

● Missed External Call: Only missed calls from trunks will be sent in email notifications.

Table 46: IAX Extension Configuration Parameters🡪Specific Time

Specific Time

Time Condition Click to add Time Condition to configure a specific time for this extension.

Follow Me

Enable Configure to enable or disable Follow Me for this user.

If the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of the
Skip Trunk Auth
Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering If enabled, call will need to be confirmed after answering.

Enable Destination Configure to enable destination.

Default Destination The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow

Use the callee DOD number as CID if configured Follow Me numbers are external numbers.
Me

Play Follow Me Prompt If enabled, the Follow Me prompt tone will be played.

Add a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan should
New Follow Me Number
have permissions to dial the defined external number.

Dialing Order This is the order in which the Follow Me destinations will be dialed to reach the user.

Create New FXS Extension

The UCM630X supports Foreign eXchange Subscriber (FXS) interface. FXS is used when a user needs to connect analog phone lines or FAX machines to the
UCM630X.

To manually create a new FXS user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new dialog window will show for users which

need to make sure first to select the extension type to be FXS Extension before proceeding to fill in the extension information. The configuration parameters
are as follows.
General

Extension The extension number associated with the user.

Analog Station Select the FXS port to be assigned for this extension.

Configure the CallerID Number that would be applied for

outbound calls from this user.

Caller ID Number Note: The ability to manipulate your outbound Caller ID may be

limited by your VoIP provider.

Assign permission level to the user. The available permissions

are “Internal”, “Local”, “National” and “International” from the

lowest level to the highest level. The default setting is “Internal”.

Privilege
Note: Users need to have the same level as or higher level than

an outbound rule’s privilege to make outbound calls using this

rule.

Configure Voicemail.

There are three valid options, and the default option is “Enable

Local Voicemail”.
Voicemail

● Disable Voicemail: Disable Voicemail.

● Enable Local Voicemail: Enable voicemail for the user.

Configure voicemail password (digits only) for the user to access

the voicemail box. A random numeric password is automatically

Voicemail Password
generated. It is recommended to use the randomly generated

password for security purposes.

Skip Voicemail When a user dials voicemail code, the password verification IVR

Password is skipped. If enabled, this would allow one-button voicemail

Verification access. By default, this option is disabled.

Send Voicemail Configures whether to send emails to the extension’s email

Email Notification address to notify of a new voicemail.

Attach Voicemail to Configures whether to attach a voicemail audio file to the

Email voicemail notification emails.

Keep Voicemail after Only applies if extension-level or global Send Voicemail to Email

Emailing is enabled.

Emergency CID CallerID name that will be used for emergency calls and

Name callbacks.
 If selected, this extension will be disabled on the UCM630X.
Disable This
Note: The disabled extension still exists on the PBX but cannot
Extension
be used on the end device.

User Settings

Configure the first name of the user. The first name can contain
First Name
characters, letters, digits, and _.

Configure the last name of the user. The last name can contain
Last Name
characters, letters, digits, and _.

Fill in the Email address for the user. Voicemail will be sent to
Email Address
this Email address.

Configure the password for user portal access. A random numeric

User Password password is automatically generated. It is recommended to use

the randomly generated password for security purposes.

Mobile Phone
Configure the Mobile number of the user.
Number

Select the voice prompt language to be used for this extension.

The default setting is “Default” which is the selected voice

prompt language under Web GUI🡪PBX Settings🡪Voice

Prompt🡪Language Settings. The dropdown list shows all the

Language
currently available voice prompt languages on the UCM630X. To

add more languages to the list, please download the voice prompt

package by selecting “Check Prompt List” under Web

GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Analog Settings

Call Waiting Configure to enable/disable call waiting feature. The default setting is “No”.

User ‘#’ as SEND If configured, the # key can be used as SNED key after dialing the number on the analog phone. The default setting is “Yes”.

RX Gain Configure the RX gain for the receiving channel of the analog FXS port. The valid range is -30dB to +6dB. The default setting is 0.

TX Gain Configure the TX gain for the transmitting channel of the analog FXS port. The valid range is -30dB to +6dB. The default setting is 0.

Configure the minimum period of time (in milliseconds) that the hook flash must remain unpressed for the PBX to consider the event
Min RX Flash
as a valid flash event. The valid range is 30ms to 1000ms. The default setting is 200ms.

Configure the maximum period of time (in milliseconds) that the hook flash must remain unpressed for the PBX to consider the event
Max RX Flash
as a valid flash event. The minimum period of time is 256ms and it cannot be modified. The default setting is 1250ms.
 If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries,
Enable Polarity
a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as Hangup on a polarity
Reversal
reversal. The default setting is “Yes”.

Specify “ON”, “OFF” or a value (the power of 2) from 32 to 1024 as the number of taps of cancellation.

Echo Cancellation Note: When configuring the number of taps, the number 256 is not translated into 256ms of echo cancellation. Instead, 256 taps mean

256/8 = 32 ms. The default setting is “ON”, which is 128 taps.

3-Way Calling Configure to enable/disable the 3-way calling feature on the user. The default setting is enabled.

Send CallerID After Configure the number of rings before sending CID. The default setting is 1.

For the FXS extension, there are three options available in Fax Mode. The default setting is “None”.

● None: Disable Fax.

● Fax Gateway: If selected, the UCM630X can support the conversation and processing of Fax data from T.30 to T.38 or T.38 to
Fax Mode T.30. only for FXS ports.

● Fax Detection: During a call, the fax signal from the user/trunk will be detected, and the received fax will be sent to the email

address configured for the user. If an email address has been configured for the user, the fax will be sent to the Default Email

Address configured in Fax/T.38->Fax Settings.

Call Transfer

Configure the Call Forward Unconditional target number. If not configured, the Call Forward Unconditional feature is
Call Forward Unconditional
deactivated. The default setting is deactivated.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The

available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or

Holiday”, and “Specific”.

Note:
CFU Time Condition
● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition
for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated.
Call Forward No Answer
The default setting is deactivated.

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”,

“Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

CFN Time Condition ● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition
for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
 Configure the Call Forward Busy target number. If not configured, the Call Forward Busy feature is deactivated. The default
Call Forward Busy
setting is deactivated.

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”,

“Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

CFB Time Condition ● “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

● Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition
for a specific time.

● Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

If Do Not Disturb is enabled, all incoming calls will be dropped.

Do Not Disturb
All call forward settings will be ignored.

DND Time Condition The time condition of DND. The DND will take effect while the time condition is satisfied.

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be

separated by new lines. Pattern matching is supported.

DND Whitelist ● Z match any digit from 1-9.

● N match any digit from 2-9.

● X match any digit from 0-9.

Calls from users in the forward whitelist will not be forwarded.

Pattern matching is supported.

FWD Whitelist ● Z match any digit from 1-9.

● N match any digit from 2-9.

● X match any digit from 0-9.

CC Settings

If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to
Enable CC
that party failed for some reason.

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for
Ring Simultaneously
outbound, the register number will be used to be displayed for the external number as the caller ID number.

External Number Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.

Time Condition for Ring

Ring the external number simultaneously along with the extension based on this time condition.
Simultaneously

Use callee DOD on FWD or

Use the callee’s DOD number as CallerID on Outgoing Forwarding or Ring Simultaneously calls.
RS
Hotline

If enabled, a hotline dialing plan will be activated, a pre-configured number will be used according to the selected Hotline
Enable Hotline
Type.

Hotline Number Configure the Hotline Number

Configure the Hotline Type:

● Immediate Hotline: When the phone is off-hook, UCM630X will immediately dial the preset number
Hotline Type
● Delay Hotline: When the phone is off hook, if there is no dialing within 5 seconds, UCM630X will dial the preset
number.

Monitor privilege control Members of the list can spy on this extension via feature codes.

Seamless transfer privilege control

Allowed to seamless transfer Members of the list can seamlessly transfer via feature code.

Other Settings

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up
(voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in

Ring Timeout the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference.

The valid range is between 5 seconds and 600 seconds.

Note: If the endpoint also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be
Auto Record
accessed under Web GUI🡪CDR🡪Recording Files.

● If set to “Yes”, users can skip entering the password when making outbound calls.
Skip Trunk Auth ● If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

● If set to “No”, users will be asked to enter the password when making outbound calls.

Time Condition for Skip If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can skip entering a password when

Trunk Auth making outbound calls.

Dial Trunk Password Configure personal password when making outbound calls via the trunk.

If enabled, this extension will be added to the LDAP Phonebook PBX list; if disabled, this extension will be skipped when
Enable LDAP
creating LDAP Phonebook.

Use MOH as IVR ringback If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of

tone the regular ringback tone.

Music On Hold Select which Music On Hold class to suggest to the extension when putting the active call on hold.
Call Duration Limit Check to enable and set the call limit the duration.

Maximum Call Duration (s) The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

Email Missed Calls Send a log of missed calls to the extension’s email address.

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

● Default: All missed calls will be sent in email notifications.

Missed Call Type
● Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

● Missed External Call: Only missed calls from trunks will be sent in email notifications.

Specific Time

Time Condition Click to add Time Condition to configure a specific time for this extension.

Follow Me

Enable Configure to enable or disable Follow Me for this user.

If the outbound calls need to check the password, we should enable this option or enable the option "Skip Trunk Auth" of the
Skip Trunk Auth
Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering If enabled, call will need to be confirmed after answering.

Enable Destination Configure to enable destination.

Default Destination The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow

Use the callee DOD number as CID if configured Follow Me numbers are external numbers.
Me

Play Follow Me Prompt If enabled, the Follow Me prompt tone will be played.

Add a new Follow Me number which could be a "Local Extension" or an "External Number". The selected dial plan should
New Follow Me Number
have permissions to dial the defined external number.

Dialing Order This is the order in which the Follow Me destinations will be dialed to reach the user.
Batch Add Extensions

Batch Add SIP Extensions

To add multiple SIP extensions, BATCH add can be used to create standardized SIP extension accounts. However, a unique extension username cannot be set

using BATCH add.

Under Web GUI🡪Extension/Trunk🡪Extensions, click on “Add” and select extension type as a SIP extension, and “Select Add Method” as Batch.

General

Create
Specify the number of extensions to be added. The default setting is 5.
Number

Extension
Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004, …… Note:
Incrementatio
Up to 3 characters.
n

Extension Configure the starting extension number of the batch of extensions to be added.

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest

level to the highest level. The default setting is “Internal”.

Permission

Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls from this rule.

Configure Voicemail.

There are three valid options, and the default option is “Enable Local Voicemail”.

Voicemail Disable Voicemail: Disable Voicemail.

Enable Local Voicemail: Enable voicemail for the user.

Enable Remote Voicemail: Forward the notify message from the remote voicemail system for the user, and the local voicemail will

be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil).

Configure the SIP/IAX password for the users. Three options are available to create a password for the batch of extensions.

User Random Password.

SIP/IAX
A random secure password will be automatically generated. It is recommended to use this password for security purposes.
Password

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.

 Configure Voicemail password (digits only) for the users.

User Random Password.

Voicemail
A random password in digits will be automatically generated. It is recommended to use this password for security purposes.
Password

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.

Send
Send voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found
Voicemail to
in Voicemail->Voicemail Email Settings.
Email

Keep

Voicemail
Only applies if extension-level or global Send Voicemail to Email is enabled.
after

Emailing

Configure CallerID Number when adding Batch Extensions.

Use Extension as Number

CallerID
Users can choose to use the extension number as CallerID
Number

Use as Number

Users can choose to set a specific number instead of using the extension number.

Skip

Voicemail When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access.

Password By default, this option is disabled.

Verification

Enable Keep-
If enabled, the PBX will regularly send SIP OPTIONS to check if the host device is online.
alive

Keep-alive
Configure the keep-alive interval (in seconds) to check if the host is up.
Frequency

Disable This
Check this box to disable this extension.
Extension

If enabled, (1) Call Forward, Call Waiting, and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1,
Enable SCA
and (3) Private numbers can be added in Call Features🡪SCA page.

Emergency
CallerID number that will be used when calling out and receiving direct callbacks.
Calls CID
 If enabled, the extension can be registered to and accessed via Wave. If disabled, Wave cannot be used, but regular registration and
Enable Wave
calling functionality will still work.

Sync Contact
If enabled, this extension number will be displayed in the Wave contact, otherwise, it will not be displayed, and it cannot be found in the

chat, but the user can still dial this number.

Select voice prompt language for this extension.

Language
If set to “Default”, the global setting for voice prompt language will be used.

Media

Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is a one-way
NAT
audio issue, usually it is related to NAT configuration or the Firewall’s support of SIP and RTP ports. The default setting is enabled.

By default, the PBX will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the
Enable Direct
endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The
Media
default setting is “No”.

Select DTMF mode for the user to send DTMF. The default setting is “RFC4733”. If “Info” is selected, the SIP INFO message will be

DTMF Mode used. If “Inband” is selected, a-law or u-law are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband”

will be used.

Alert-info When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.

SRTP Enable/disable SRTP for RTP stream encryption.

Configure to enable Packet Loss Retransmission.

Packet Loss
NACK
Retransmissio

n NACK+RTX(SSRC-GROUP)

OFF

Video FEC Check to enable Forward Error Correction (FEC) for Video.

FECC Configure to enable FECC

Audio FEC Check to enable Forward Error Correction (FEC) for Audio.

Access Control List manages the IP addresses that can register to this extension.

Allow All: Any IP address can register to this extension.

ACL Policy

Local Network: Only IP addresses in the configured network segments can register to this extension. Press “Add Local Network

Address” to add more IP segments.

 Select the jitter buffer method.

Disable: Jitter buffer will not be used.

Jitter Buffer Fixed: Jitter buffer with a fixed size (equal to the value of “jitter buffer size”)

Adaptive: Jitter buffer with an adaptive size (no more than the value of “max jitter buffer”).

NetEQ: Dynamic jitter buffer via NetEQ.

Codec
Configure the codecs to be used.
Preference

Call Transfer

Presence Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are
Status possible: “Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”. More details at [PRESENCE].

Enable and configure the Call Forward Unconditional target number. Available options for target number are:

“None”: Call forward deactivated.

“Extension”: Select an extension from the dropdown list as CFU target.

“Custom Number”: Enter a customer number as a target. For example: *97.

Call Forward “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected

Unconditional extension.

“Ring Group”: Select a ring group from the dropdown list as CFU target.

“Queues”: Select a queue from the dropdown list as CFU target.

“Voicemail Group”: Select a voicemail group from the dropdown list as CFU target.

The default setting is “None”.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time

conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Note:
CFU Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
 Configure the Call Forward No Answer target number. Available options for target number are:

“None”: Call forward deactivated.

“Extension”: Select an extension from the dropdown list as CFN target.

“Custom Number”: Enter a customer number as a target. For example: *97.

Call Forward “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected

No Answer extension.

“Ring Group”: Select a ring group from the dropdown list as CFN target.

“Queues”: Select a queue from the dropdown list as CFN target.

“Voicemail Group”: Select a voicemail group from the dropdown list as CFN target.

The default setting is “None”.

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”,

“Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
CFN Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Configure the Call Forward Busy target number. Available options for target number are:

“None”: Call forward deactivated.

“Extension”: Select an extension from the dropdown list as CFB target.

“Custom Number”: Enter a customer number as a target. For example: *97.

Call Forward “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected

Busy extension.

“Ring Group”: Select a ring group from the dropdown list as CFB target.

“Queues”: Select a queue from the dropdown list as CFB target.

“Voicemail Group”: Select a voicemail group from the dropdown list as CFB target.

The default setting is “None”.

 Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
CFB Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

If Do Not Disturb is enabled, all incoming calls will be dropped.

Do Not

Disturb
All call forward settings will be ignored.

Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

DND Time
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.
Condition

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a
specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new

lines. Pattern matching is supported.

DND
Z matches any digit from 1-9
Whitelist
N matches any digit from 2-9

X matches any digit from 0-9.

Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

FWD Z matches any digit from 1-9

Whitelist
N matches any digit from 2-9

X matches any digit from 0-9.

CC Settings

If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party
Enable CC
failed for some reason. By default, it is disabled.
 Two modes for Call Completion are supported:

Normal: This extension is used as an ordinary extension.

CC Mode
For Trunk: This extension is registered from a PBX.

The default setting is “Normal”.

CC Max Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the
Agents maximum number of CC requests this channel can make. The minimum value is 1.

CC Max Configure the maximum number of monitor structures that may be created for this device. In other words, this number tells how many

Monitors callers may request CC services for a specific device at one time. The minimum value is 1.

Ring

Simultaneousl

Ring
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the
Simultaneousl
register number will be used to be displayed for the external number as the caller ID number.
y

Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.
External

Number
This field accepts only letters, numbers, and special characters + = * #.

Time

Condition for

Ring Ring the external number simultaneously along with the extension based on this time condition.
Simultaneousl

Use callee

DOD on Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.

FWD or RS

Monitor
privilege

control

Allowed to
Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using

feature code.
call-barging

Seamless

transfer

privilege

control
Allowed to
Any extensions on the UCM can perform a seamless transfer. When using the Pickup Incall feature, only extensions available on the
seamless
“Selected Extensions” list can perform a seamless transfer to the edited extension.
transfer

Other
Settings

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout

setting under Web GUI 🡪 PBX Settings 🡪 Voice Prompt 🡪 Custom Prompt: General Preference. The valid range is between 3 seconds
Ring Timeout
and 600 seconds.

Note: If the endpoint also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web
Auto Record
GUI🡪CDR🡪Recording Files.

If set to “yes”, users can skip entering the password when making outbound calls.
Skip Trunk
If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.
Auth
If set to “No”, users will be asked to enter the password when making outbound calls.

Time
Condition for If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering a password when making outbound

Skip Trunk calls.

Auth

Dial Trunk
Configure personal password when making outbound calls via the trunk.
Password

Enable LDAP If enabled, the extension will be added to the LDAP Phonebook PBX list.

Bind PMS If enabled, the system will create a room whose room number, by default, will equal the extension number in the PMS module. Note: If

Room this room already exists, the configuration of the existing room will be overwritten.

Music On
Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Hold

Call Duration
The maximum duration of call-blocking.
Limit

Maximum
The maximum call duration (in seconds). The default value 0 means no limit.
Call Duration

If disabled, UCM will not invite the extension when it is already in a call and will do the same work as the user is busy.
Call Waiting
Note: the option only works when the caller dials the extension directly.
 Table 53: Batch Add SIP Extension Parameters

Batch Add IAX Extensions

Under Web GUI🡪Extension/Trunk🡪Extensions, click on “Add”, then select extension type as IAX Extension and the add method to be Batch.

General

Create
Specify the number of extensions to be added. The default setting is 5.
Number

Extension

Incrementatio Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004, ……

Extension The extension number associated with this particular user/phone.

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest

level to the highest level. The default setting is “Internal”.

Permission

Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls from this rule.

CallerID Configure the Caller ID number displayed when dialing calls from this user. Note: The Caller ID usage might be limited by your VoIP

Number provider. In Batch Add Method, “e” means to use the extension as the number.

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

Voicemail Disable Voicemail: Disable Voicemail.

Enable Local Voicemail: Enable voicemail for the user.

Configure the SIP/IAX password for the users. Three options are available to create a password for the batch of extensions.

User Random Password.

SIP/IAX
A random secure password will be automatically generated. It is recommended to use this password for security purposes.
Password

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.

Configure Voicemail password (digits only) for the users.

User Random Password.

Voicemail
A random password in digits will be automatically generated. It is recommended to use this password for security purposes.
Password

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.

Send
Send voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found
Voicemail to
in Voicemail 🡪 Voicemail Email Settings.
Email

Keep
Voicemail
Only applies if extension-level or global Send Voicemail to Email is enabled.
after
Emailing

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under
Auto Record
Web GUI🡪CDR🡪Recording Files.

Skip

Voicemail When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access.

Password By default, this option is disabled.

Verification

Disable This
Check this box to disable this extension.
Extension

Language Select voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.

IAX Settings

Max Number
Configure the maximum number of calls allowed for each remote IP address.
of Calls

Configure to enable/disable requiring call token. If set to “Auto”, it might lock out users who depend on backward compatibility when
Require Call peer authentication credentials are shared between physical endpoints.
Token
The default setting is “Yes”.

SRTP Enable/disable SRTP for RTP stream encryption.

Access Control List manages the IP addresses that can register to this extension.

ACL Policy Allow All: Any IP address can register to this extension.

Local Network: Only IP addresses in the configured network segments can register to this extension.

Codec
Configure the codecs to be used.
Preference

Call Transfer

Call Forward
Enable and configure the Call Forward Unconditional target number.
Unconditional
 Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time
conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Note:
CFU Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward
Configure the Call Forward No Answer target number.
No Answer

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”,

“Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
CFN Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a
specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward
Configure the Call Forward Busy target number.
Busy

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
CFB Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

If Do Not Disturb is enabled, all incoming calls will be dropped.

Do Not

Disturb
All call forward settings will be ignored.
 Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:
DND Time

Condition “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a

specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by
new lines. Pattern matching is supported.

DND
Z match any digit from 1-9，
Whitelist
N match any digit from 2-9，

X match any digit from 0-9.

Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

FWD Z match any digit from 1-9，

Whitelist
N match any digit from 2-9，

X match any digit from 0-9.

Ring

Simultaneousl

Ring
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the
Simultaneousl
register number will be used to be displayed for the external number as the caller ID number.
y

Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.
External

Number
This field accepts only letters, numbers, and special characters + = * #.

Time

Condition for
Ring Ring the external number simultaneously along with the extension based on this time condition.
Simultaneousl
y

Use callee
DOD on FWD Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.

or RS
Monitor
privilege

control

Allowed to
Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using
feature code.
call-barging

Seamless

transfer

privilege

control

Allowed to
Any extensions on the UCM can perform a seamless transfer. When using the Pickup Incall feature, only extensions available on the
seamless
“Selected Extensions” list can perform a seamless transfer to the edited extension.
transfer

Other
Settings

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout

setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference. The valid range is between 5 seconds and
Ring Timeout
600 seconds.

Note: If the endpoint also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web
Auto Record
GUI🡪CDR🡪Recording Files.

If set to “yes”, users can skip entering the password when making outbound calls.
Skip Trunk
If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.
Auth
If set to “No”, users will be asked to enter the password when making outbound calls.

Time
Condition for If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering a password when making outbound

Skip Trunk calls.

Auth

Dial Trunk
Configure personal password when making outbound calls via the trunk.
Password

Enable LDAP If enabled, the extension will be added to the LDAP Phonebook PBX list.

Music On
Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Hold
 Call Duration
Check to enable and set the call limit the duration.
Limit

Maximum

Call Duration The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

(s)

Table 54: Batch Add IAX Extension Parameters

Batch Extension Resetting Functionality

Users can select multiple extensions and reset their settings to default by pressing the reset button and confirming the reset functionality. Once

done, all settings on the Basic Settings page will be restored to default values except for Concurrent Registrations. User voicemail password will be reset to

Random Password. User voicemail prompts and recordings will be deleted. User Management settings will also be restored to default except for usernames

and custom privileges

Search and Edit Extension

All the UCM630X extensions are listed under Web GUI🡪Extension/Trunk🡪Extensions, with status, Extension, CallerID Name, Technology (SIP, IAX, and

FXS), IP and Port. Each extension has a checkbox for users to “Edit” or “Delete”. Also, options “Edit” , “Reboot” and “Delete” are available per
extension. Users can search for an extension by specifying the extension number to find an extension quickly.

Manage Extensions

Status

Users can see the following icon for each extension to indicate the SIP status.

Green: Idle
 Blue: Ringing

Yellow: In Use

Grey: Unavailable (the extension is not registered or disabled on the PBX)

Edit single extension

Click on to start editing the extension parameters.

Reset single extension

Click on to reset the extension parameters to default (except concurrent registration).

Other settings will be restored to default in Maintenance🡪User Management🡪User Information except for username and permissions and delete the user
voicemail prompt and voice messages.

Note

This is the expected behavior when you reset an extension:

All the data and configuration on the user side will be deleted. That includes user information, call history, call recordings, faxes, voice mails, meeting

schedules and recordings, as well as chat history. However, the data related to the user will be kept on the UCM side.

The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search

through those messages while the new user of the extension cannot.

If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about

the meeting.

Reboot the user

Click on to send NOTIFY reboot event to the device which has a UCM630X extension already registered. To successfully reboot the user, “Zero Config”

needs to be enabled on the UCM630X Web GUI🡪Other Features🡪Zero Config🡪Zero Config Settings.

Delete single extension

Click on to delete the extension. Or select the checkbox of the extension and then click on “Delete Selected Extensions”.

Notes

This is the expected behavior when you delete an extension:

The system will delete all the data of the extension except the CDR and meetings record. All the data on the user side will be erased.

The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search

through those messages while the new user of the extension cannot.

If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about

the meeting.

Modify selected extensions

Select the checkbox for the extension(s). Then click on “Edit” to edit the extensions in a batch.

Delete selected extensions

Select the checkbox for the extension(s). Then click on “Delete ” to delete the extension(s).

Export Extensions
The extensions configured on the UCM630X can be exported to a CSV format file with selected technology “SIP”, “IAX” or “FXS”. Click on the “Export
Extensions” button and select technology in the prompt below.

Export Extensions

Export Basic Settings

Export Basic Information includes:

Extension

CallerID Number

Privilege

SIP/IAX Password

AuthID

Voicemail

Voicemail Password

Sync Contact

First Name

Last Name
 Email Address

User/Wave Password

If importing extensions with no values for settings, the following will occur:

If importing new extensions, or if Replace is selected as the duplicate import option, the default values for those settings will be used.

If Update is selected as the duplicate import option, no changes will be made to the existing settings.

The exported CSV file can serve as a template for users to fill in desired extension information to be imported to the UCM630X.

Import Extensions
The capability to import extensions to the UCM630X provides users the flexibility to batch add extensions with similar or different configurations quickly
into the PBX system.

1. Export extension CSV file from the UCM630X by clicking on the “Export Extensions” button.

2. Fill up the extension information you would like in the exported CSV template.

3. Click on the “Import Extensions” button. The following dialog will be prompted.

Import Extensions

4. Select the option in “On Duplicate Extension” to define how the duplicate extension(s) in the imported CSV file should be treated by the PBX.

Skip: Duplicate extensions in the CSV file will be skipped. The PBX will keep the current extension information as previously configured without

change.

Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the CSV file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the CSV file has a

different configuration for any options, it will override the configuration for those options in the extension.

5. Click on “Choose file to upload” to select a CSV file from a local directory on the PC.

6. Click on “Apply Changes” to apply the imported file on the UCM630X.

Example of a file to import:

Import File
Field Supported Values

Extension Digits

Technology SIP/SIP(WebRTC)

Enable Voicemail yes/no/remote

CallerID Number Digits

SIP/IAX Password Alphanumeric characters

Voicemail Password Digits

Skip Voicemail Password

yes/no
Verification

Ring Timeout Empty/ 3 to 600 (in second)

SRTP yes/no

Strategy Allow All/Local Subnet Only/A Specific IP Address

Local Subnet 1 IP address/Mask

Local Subnet 2 IP address/Mask

Local Subnet 3 IP address/Mask

Local Subnet 4 IP address/Mask

Local Subnet 5 IP address/Mask

Local Subnet 6 IP address/Mask

Local Subnet 7 IP address/Mask

Local Subnet 8 IP address/Mask

Local Subnet 9 IP address/Mask

Local Subnet 10 IP address/Mask

Specific IP Address IP address

Skip Trunk Auth yes/no/bytime

Field Supported Values

Codec Preference PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Permission Internal/Local/National/International

NAT yes/no

DTMF Mode RFC4733/info/inband/auto

Insecure Port

Enable Keep-alive Yes/no

Keep-alive Frequency Value from 1-3600

AuthID Alphanumeric value without special characters

TEL URI Disabled/user=phone/enabled

Call Forward Busy Digits

Call Forward No Answer Digits

Call Forward Unconditional Digits

Support Hot-Desking Mode Yes/no

Dial Trunk Password Digits

Disable This Extension Yes/no

CFU Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFB Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold Default/ringbacktone_default

If CC is disabled use: never

CC Agent Policy If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor Policy Generic/never

Field Supported Values

CCBS Available Timer 3600/4800

CCNR Available Timer 3600/7200

CC Offer Timer 60/120

CC Max Agents Value from 1-999

CC Max Monitors Value from 1-999

Ring simultaneously Yes/no

External Number Digits

Time Condition for Ring

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Simultaneously

Time Condition for Skip Trunk

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Auth

Enable LDAP Yes/no

Enable T.38 UDPTL Yes/no

Max Contacts Values from 1-10

Enable Wave Yes/no

None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-
Alert-Info
dr3/ bellcore-dr4/ bellcore-dr5/custom

Do Not Disturb Yes/no

DND Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Custom Auto answer Yes/no

Do Not Disturb Whitelist Empty/digits

User Password Alphanumeric characters.

First Name Alphanumeric without special characters.

Last Name Alphanumeric without special characters.

Field Supported Values

Email Address Email address

Language Default/en/zh

Phone Number Digits

Call-Barging Monitor Extensions allowed to call barging

Seamless Transfer Members Extensions allowed to seamless transfer

Table 55: SIP extensions Imported File Example

Field Supported Values

Extension Digits

Technology IAX

Enable Voicemail yes/no

CallerID Number Digits

SIP/IAX Password Alphanumeric characters

Voicemail Password Digits

Skip Voicemail Password Verification yes/no

Ring Timeout Empty/ 3 to 600 (in second)

SRTP yes/no

Strategy Allow All/Local Subnet Only/A Specific IP Address

Local Subnet 1 IP address/Mask

Local Subnet 2 IP address/Mask

Local Subnet 3 IP address/Mask

Local Subnet 4 IP address/Mask

Local Subnet 5 IP address/Mask

Local Subnet 6 IP address/Mask

Field Supported Values

Local Subnet 7 IP address/Mask

Local Subnet 8 IP address/Mask

Local Subnet 9 IP address/Mask

Local Subnet 10 IP address/Mask

Specific IP Address IP address

Skip Trunk Auth yes/no/bytime

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-
Codec Preference
32,ADPCM,G.723,H.263,H.263p,vp8,opus

Permission Internal/Local/National/International

NAT yes/no

Call Forward Busy Digits

Call Forward No Answer Digits

Call Forward Unconditional Digits

Require Call Token Yes/no/auto

Max Number of Calls Values from 1-512

Dial Trunk Password Digits

Disable This Extension Yes/no

CFU Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

CFB Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold Default/ringbacktone_default

Ring simultaneously Yes/no

External Number Digits

Field Supported Values

Time Condition for Ring

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Simultaneously

Time Condition for Skip Trunk Auth All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP Yes/no

Limit Max time (s) empty

Do Not Disturb Yes/no

DND Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Do Not Disturb Whitelist Empty/digits

User Password Alphanumeric characters.

First Name Alphanumeric without special characters.

Last Name Alphanumeric without special characters.

Email Address Email address

Language Default/en/zh

Phone Number Digits

Call-Barging Monitor Extensions allowed to call barging

Seamless Transfer Members Extensions allowed to seamless transfer

Table 56: IAX extensions Imported File Example

Field Supported Values

Extension Digits

Technology FXS

Analog Station FXS1/FXS2

Enable Voicemail yes/no

CallerID Number Digits

Voicemail Password Digits

Field Supported Values

Skip Voicemail Password Verification yes/no

Ring Timeout Empty/ 3 to 600 (in second)

Auto Record yes/no

Fax Mode None/Fax Gateway/Fax Detection

Skip Trunk Auth Yes/no/bytime

Permission Internal/Local/National/International

Call Forward Busy Digits

Call Forward No Answer Digits

Call Forward Unconditional Digits

Call Waiting Yes/no

Use # as SEND Yes/no

RX Gain Values from -30🡪6

TX Gain Values from -30🡪6

MIN RX Flash Values from: 30 – 1000

MAX RX Flash Values from: 40 – 2000

Enable Polarity Reversal Yes/no

Echo Cancellation On/Off/32/64/128/256/512/1024

3-Way Calling Yes/no

Send CallerID After 1/2

Dial Trunk Password digits

Disable This Extension Yes/no

CFU Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Field Supported Values

CFB Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold Default/ringbacktone_default

If CC is disabled use: never

CC Agent Policy If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor Policy Generic/never

CCBS Available Timer 3600/4800

CCNR Available Timer 3600/7200

CC Offer Timer 60/120

CC Max Agents Value from 1-999

CC Max Monitors Value from 1-999

Ring simultaneously Yes/no

External Number Digits

Time Condition for Ring Simultaneously All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

Enable LDAP Yes/no

Enable Hotline Yes/no

Hotline Type Immediate hotline/delay hotline

Hotline Number digits

Limit Max time (s) empty

Do Not Disturb Yes/no

DND Time Condition All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Do Not Disturb Whitelist Empty/digits

 Field Supported Values

User Password Alphanumeric characters.

First Name Alphanumeric without special characters.

Last Name Alphanumeric without special characters.

Email Address Email address

Language Default/en/zh

Phone Number Digits

Call-Barging Monitor Extensions allowed to call barging

Seamless Transfer Members Extensions allowed to seamless transfer

Table 57: FXS Extensions Imported File Example

The CSV file should contain all the above fields, if one of them is missing or empty, the UCM630X will display the following error message for missing

fields.

Import Error

Extension Details
Users can click on an extension number in the Extensions list page and quickly view information about it such as:

Extension: This shows the Extension number.

Status: This shows the status of the extension.

Presence status: Indicates the Presence Status of this extension.

Terminal Type: This shows the type of the terminal using this extension (SIP, FXS…etc.).

Caller ID Name: Reveals the Caller ID Name configured on the extension.

Messages: Shows the messages’ stats.

IP and Port: The IP address and the ports of the device using the extension.

Email status: Show the Email status (sent, to be sent…etc.).

Ring Group: Indicates the ring groups that this extension belongs to.

Call Queue: Indicates the Cal Queues that this extension belongs to.

Call Queue (Dynamic): Indicates the Call Queues that this extension belongs to as a dynamic agent.
 Extension Details

E-mail Notification
Once the extensions are created with Email addresses, the PBX administrator can click on the button “E-mail Notification” to send the account registration

and configuration information to the user. Please make sure the Email setting under Web GUI🡪System Settings🡪Email Settings is properly configured and
tested on the UCM630X before using “E-mail Notification”.

When clicking on ”More” > “E-mail Notification” button, the following message will be prompted on the web page. Click on OK to confirm sending the

account information to all users’ Email addresses.

E-mail Notification – Prompt Information

The user will receive an Email including account registration information as well as the Wave Settings with the QR code:
 Wave Settings and QR Code

Important Note

For security and confidentiality reasons, it is high advisable for the user to change the Wave login extension after the first time log in.

The UCM admin can also send “Remote Registration Mail” and “Wave Welcome Mail” as the figure below shows

Send Remote Registration Mail

Multiple Registrations per Extension

UCM630X supports multiple registrations per extension so that users can use the same extension on devices in different locations.
 Multiple Registrations per Extension

This feature can be enabled by configuring the option “Concurrent Registrations” under Web GUI🡪Extension/Trunk🡪Edit Extension. The default value is
set to 1 for security purposes. The maximum is 10.

Extension – Concurrent Registration

SMS Message Support

The UCM630X provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that support SIP message, after a

UCM630X account is registered on the end device, the user can send and receive SMS messages. Please refer to the end device documentation on how to send

and receive SMS messages.

SMS Message Support

EXTENSION GROUPS
The UCM630X extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the

UCM630X. For example, when configuring “Enable Filter on Source Caller ID”, users could select a group instead of each person’s extension to assign. This
feature simplifies the configuration process and helps manage and categorize the extensions for a business environment.

Configure Extension Groups

Extension groups can be configured via Web GUI🡪Extension/Trunk🡪Extension Groups.

Click on to create a new extension group.

Click on to edit the extension group.

Click on to delete the extension group.

Select extensions from the list on the left side to the right side.

Edit Extension Group

Click on to change the ringing priority of the members selected on the group.

Using Extension Groups

Here is an example where the extension group can be used. Go to Web GUI🡪Extension/Trunk🡪Outbound Routes and select “Enable Filter on Source

Caller ID”. Both single extensions and extension groups will show up for users to select.

Select Extension Group in Outbound Route

ANALOG TRUNKS
Go to Web GUI🡪Extension/Trunk🡪Analog Trunks to add and edit analog trunks.

Click on “Create New Analog Trunk” to add a new analog trunk.

Click on to edit the analog trunk.

Click on to delete the analog trunk.

Analog Trunk Configuration

The analog trunk options are listed in the table below.

Select the channel for the analog trunk.

UCM6301: 1 channel

FXO Port UCM6302: 2 channels

UCM6304: 4 channels

UCM6308: 8 channels

Trunk Name Specify a unique label to identify the trunk when listed in outbound rules, incoming rules, etc.

Advanced
Options

Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a
SLA Mode
PBX. Enable SLA Mode will disable polarity reversal.

The barge option specifies whether other stations can join a call in progress on this trunk. If enabled, the other stations can press the line

button to join the call.

Barge Allowed

The default setting is Yes.

The hold option specifies hold permissions for this trunk. If set to “Open”, any station can place this trunk on hold and any other station

is allowed to retrieve the call. If set to “Private”, only the station that places the call on hold can retrieve the call.
Hold Access

The default setting is Yes.

Enable If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a
Polarity polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as “Hangup” on a polarity

Reversal reversal. The default setting is “No”.

Polarity on When the FXO port answers the call, FXS may send a Polarity Reversal. If this interval is shorter than the value of “Polarity on Answer
Answer Delay Delay”, the Polarity Reversal will be ignored. Otherwise, the FXO will On-hook to disconnect the call. The default setting is 600ms.
Current
Disconnect This is the periodic time (in ms) that the UCM630X will use to check on a voltage drop in the line. The default setting is 200. The valid

Threshold range is 50 to 3000.

(ms)

Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a Hangup before the line is

Ring Timeout answered. This value can be used to configure how long it takes before the UCM630X considers a non-ringing line with Hangup

activity. The default setting is 8000.

Configure the RX gain for the receiving channel of the analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default
RX Gain
setting is 0.

Configure the TX gain for the transmitting channel of the analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The
TX Gain
default setting is 0.

Configure to enable CallerID detection.

Use CallerID
The default setting is “Yes”.

Select the Caller ID scheme for this trunk.

Bellcore/Telcordia.

ETSI-FSK During Ringing

ETSI-FSK Prior to Ringing with DTAS

ETSI-FSK Prior to Ringing with LR

ETSI-FSK Prior to Ringing with RP

Caller ID ETSI-DTMF During Ringing

Scheme ETSI-DTMF Prior to Ringing with DTAS

ETSI-DTMF Prior to Ringing with LR

ETSI-DTMF Prior to Ringing with RP

SIN 227-BT

NTT Japan

Auto-Detect

If you are not sure which scheme to choose, please select “Auto Detect”. The default setting is “Bellcore/Telcordia”.

Configures how faxes to this extension will be handled.

None: Faxes will not be processed.

Fax Mode
Fax Gateway: Faxes to this extension will be processed and converted from T.30 to T.38 or vice-versa. FXS/FXO ports only.

The default setting is None.

FXO Dial
Configure the time interval between off-hook and first dialed digit for outbound calls.
Delay (ms)

Enable automatic recording for the calls using this trunk. The default setting is disabled. The recording files can be accessed under Web
Auto Record
GUI🡪CDR🡪Recording Files.

Disable This
If selected, the trunk will be disabled, and incoming/Outgoing calls via this trunk will not be possible.
Trunk

This is to implement an analog trunk outbound line selection strategy.

Three options are available:

Ascend

When the call goes out from this analog trunk, it will always try to use the first idle FXO port. The port order that the call will use to go

out if UCM6302 is used would be port 1🡪port 2🡪. Every time it will start with port 1 (if it is idle).

DAHDI Out Poll

Line Selection
When the call goes out from this analog trunk, it will use the port that is not used last time. And it will always use the port in the order of

port 1🡪2🡪1🡪2🡪1🡪2🡪…, following the last port being used in case UCM6302, is used.

Descend

When the call goes out from this analog trunk, it will always try to use the last idle FXO port. The port order that the call will use to go
out if UCM6302 is used would be port 2🡪port 1. Every time it will start with port 2 (if it is idle).

The default setting is “Ascend” mode.

The Non-Linear Processing (NLP) in echo cancellation helps to remove/suppress residual echo components that could not be removed
by the LEC (Line Echo Canceller). Following modes are supported:

Default: The NLP limits the signal level to the background noise level when active, and the background noise level adjustment is

Echo low.

Cancellation High Noise Level Adjustment: The NLP limits the signal level to the background noise level when active, and the background
Mode noise level adjustment is high.

Noise Masking: The NLP sends sign noise when active, and the background noise level adjustment is high.

White Noise Injection: The NLP injects white noise when active. The level corresponds to the background noise level at Sin, and

the background noise level adjustment is high.

Allows external numbers the option to get directed to the extension that last called them.

Direct
For Example, User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call
Callback
notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly

bridged to user 2002 regardless of the configured inbound destination.

Tone Settings
Busy
Busy Detection is used to detect far-end Hangup or for detecting busy signal. The default setting is “Yes”.
Detection

If “Busy Detection” is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2.
Busy Tone
Better results might be achieved if set to 4, 6, or even 8. Please note that the higher the number is, the more time is needed to Hangup the
Count
channel. However, this might lower the probability to get a random Hangup.

Congestion
Congestion detection is used to detect far-end congestion signal. The default setting is “Yes”.
Detection

Congestion
If “Congestion Detection” is enabled, users can specify the number of congestion tones to wait for. The default setting is 2.
Count

Select the country for tone settings. If “Custom” is selected, users could manually configure the values for Busy Tone and Congestion
Tone Country
Tone. The default setting is “United States of America (USA)”.

Syntax:

f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];

Frequencies are in Hz and cadence on and off are in ms.

Frequencies Range: [0, 4000)

Busy Tone Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

Select Tone Country “Custom” to manually configure Busy Tone value.

Default value:

f1=480@-50,f2=620@-50,c=500/500

Syntax:

f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];

Frequencies are in Hz and cadence on and off are in ms.

Frequencies Range: [0, 4000)

Congestion
Busy Level Range: (-300, 0)
Tone

Cadence Range: [0, 16383].

Select Tone Country “Custom” to manually configure Busy Tone value.

Default value:

f1=480@-50,f2=620@-50,c=250/250
 Click on “Detect” to detect the busy tone, Polarity Reversal, and Current Disconnect by PSTN. Before the detection, please make sure
PSTN
there is more than one channel configured and working properly. If the detection has a busy tone, the “Tone Country” option will be set
Detection
as “Custom”.

Table 58: Analog Trunk Configuration Parameters

PSTN Detection
The UCM630X provides a PSTN detection function to help users detect the busy tone, Polarity Reversal, and Current Disconnect by making a call from the

PSTN line to another destination. The detecting call will be answered and up for about 1 minute. Once done, the detecting result will show and can be used for

the UCM630X settings.

1. Go to UCM630X Web GUI🡪Extension/Trunk🡪Analog Trunks page.

2. Click to edit the analog trunk created for the FXO port.

3. In the window to edit the analog trunk, go to the “Tone Settings” section and there are two methods to set the busy tone.

Tone Country. The default setting is “United States of America (USA)”.

PSTN Detection.

UCM630X FXO Tone Settings

4. Click on “Detect” to start PSTN detection.

UCM630X PSTN Detection

If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.

Detect Model: Auto Detect.

Source Channel: The source channel to be detected.

Destination Channel: The channel to help detect. For example, the second FXO port.
 Destination Number: The number to be dialed for detecting. This number must be the actual PSTN number for the FXO port used as the destination

channel.

UCM630X PSTN Detection: Auto-Detect

If there is only one FXO port connected to the PSTN line, use the following settings for auto-detection.

UCM630X PSTN Detection: Semi-Auto Detect

Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

Destination Number: The number to be dialed for detecting. This number could be a cell phone number or another PSTN number that can be reached

from the source channel PSTN number.

5. Click “Detect” to start detecting. The source channel will initiate a call to the destination number. For “Auto Detect”, the call will be automatically

answered. For “Semi-auto Detect”, the UCM630X Web GUI will display a prompt to notify the user to answer or hang up the call to finish the detecting
process.

6. Once done, the detected result will show. Users could save the detecting result as the current UCM630X settings.

Detect Model Select “Auto Detect” or “Semi-auto Detect” for PSTN detection.

● Auto-Detect: Please make sure two or more channels are connected to the UCM630X and in idle status before starting

the detection. During the detection, one channel will be used as the caller (Source Channel) and another channel will be

used as a callee (Destination Channel). The UCM630X will control the call to be established and hang up between caller

and callee to finish the detection.

 ● Semi-auto Detect: Semi-auto detection requires answering or hanging up the call manually. Please make sure one

channel is connected to the UCM630X and in idle status before starting the detection. During the detection, the source-

channel will be used as a caller and send the call to the configured Destination Number. Users will then need to follow

the prompts in Web GUI to help finish the detection.

The default setting is “Auto Detect”.

Source Channel Select the channel to be detected.

Destination Channel Select the channel to help detect when “Auto Detect” is used.

Destination Number Configure the number to be called to help the detection.

Dump Call Progress Tone File Choose whether to save the calling tone file, it is not checked by default.

Table 59: PSTN Detection for Analog Trunk

The PSTN detection process will keep the call up for about 1 minute.

If “Semi-auto Detect’ is used, please pick up the call only after being informed from the Web GUI prompt.

Once the detection is successful, the detected parameters “Busy Tone”, “Polarity Reversal” and “Current Disconnect by PSTN” will be filled into the
corresponding fields in the analog trunk configuration.

VOIP TRUNKS

VoIP Trunk Configuration

VoIP trunks can be configured in UCM630X under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Once created, the VoIP trunks will be listed with Provider
Name, Type, Hostname/IP, Username, and Options to edit/detect the trunk.

Click on “Add SIP Trunk” or “Add IAX Trunk” to add a new VoIP trunk.

Click on to configure detailed parameters for the VoIP trunk.

Click on to configure Direct Outward Dialing (DOD) for the SIP Trunk.

Click on to start LDAP Sync.

Click on to delete the VoIP trunk.

The VoIP trunk options are listed in the table below.

Select the VoIP trunk type.

Type ● Peer SIP Trunk

● Register SIP Trunk

Provider Name Configure a unique label (up to 64 characters) to identify this trunk when listed in outbound rules, inbound rules, etc.

Host Name Configure the IP address or URL for the VoIP provider’s server of the trunk.
 Configure the SIP Transport method.

Using TCP requires local TCP support.

Using TLS Requires local TLS support.

Transport
● UDP

● TCP

● TLS

Auto Record If enabled, calls handled with this extension/trunk will automatically be recorded.

Keep the CID from the inbound call when dialing out. This setting will override the “Keep Trunk CID” option. Please make
Keep Original CID
sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line.

If enabled, the trunk CID will not be overridden by the extension’s CID when the extension has CID configured. The default
Keep Trunk CID
setting is “No”.

Turn on this setting when the PBX is using public IP and communicating with devices behind NAT. If there is a one-way audio
NAT
issue, usually it is related to NAT configuration or SIP/RTP port support on the firewall.

If checked, the trunk will be disabled.

Disable This Trunk Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the

SIP provider.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone"

TEL URI parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to

"Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it

might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will
Caller ID Number be set before sending out the call:

From the user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID

(Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled)

🡪 Global Outbound CID.

CallerID Name Configure the new name of the caller when the extension has no CallerID Name configured.

Auto Record If enabled, calls handled with this extension/trunk will automatically be recorded.

Auth ID Enter the Authentication ID for the "Register SIP Trunk" type.

Allows external numbers the option to get directed to the extension that last called them.

For Example, User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed
Direct Callback
call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they

will be directly bridged to user 2002 regardless of the configured inbound destination.

If enabled, the RemoteConnect related parameters will be set synchronously. Please make sure the trunk host is allocated by
RemoteConnect Mode
GDMS or supports TLS.
 If enabled and when the number of concurrent calls exceeds any trunk's configured concurrent call thresholds, an alarm
Limit Concurrent Calls
notification will be generated. Note: Please make sure the system alert event "Trunk Concurrent Calls" is enabled.

Concurrent Call Threshold Threshold of all incoming and outgoing concurrent calls through this trunk.

Outgoing Concurrent Calls

Threshold of all outgoing concurrent calls passing through this trunk.
Threshold

Incoming Concurrent Calls

Threshold of all incoming concurrent calls passing through this trunk.
Threshold

Total Time Limit For Outbound Calls

Enable Total Time Limit For

When this setting is activated, the user can set a time limit before calls cannot be initiated through this trunk
Outgoing Calls

This setting defines how long until the time allowed for outgoing calls is reset.

● Monthly: The time allowed will reset every month.

Period
● Quarterly: The time allowed will reset every 3 months.

Example: If the time limit has been set to 4320 minutes, the allowed time will always revert back to 4320 after a month or 3

month based on the period configured.

Total Time Total time allowed in minutes

SIP Peer Trunk SIP Register Trunk

Basic Settings

Provider Name Configure a unique label to identify this trunk when listed in outbound rules, inbound rules, etc.

Host Name Configure the IP address or URL for the VoIP provider’s server of the trunk.

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording
Auto Record
files can be accessed under Web GUI🡪CDR🡪Recording Files.

Keep the CID from the inbound call when dialing out, this setting will override the “Keep Trunk CID” option. Please make
Keep Original CID
sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line.

If enabled, the trunk CID will not be overridden by the extension’s CID when the extension has CID configured. The default
Keep Trunk CID
setting is “No”.

Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is a one-way audio
NAT
issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall.
 If selected, the trunk will be disabled.

Disable This Trunk Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the

SIP provider.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter

TEL URI will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:"

will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it

might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will
Caller ID Number be set before sending out the call:

● CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD

🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.

CallerID Name Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Configure the SIP transport protocol to be used in this trunk. The default setting is "UDP".

● UDP
Transport
● TCP

● TLS

RemoteConnect If enabled, the RemoteConnect related parameters will be set synchronously. Please make the trunk host is allocated by GDMS

Mode or it supports TLS.

Allows external numbers the option to get directed to the extension that last called them.

For Example, User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed
Direct Callback
call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they

will be directly bridged to user 2002 regardless of the configured inbound destination.

Advanced Settings

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726,
Codec Preference
G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8.

Packet Loss
Configure to enable Packet Loss Retransmission.
Retransmission

Audio FEC Configure to enable Forward Error Correction (FEC) for audio.

Video FEC Configure to enable Forward Error Correction (FEC) for video.

ICE Support Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end.

TURN Relay TURN servers are used for media NAT traversal and will be prioritized if ICE is also enabled.
FECC Configure to enable Far-end Camera Control

If enabled, the UCM will send CN packets for silence suppression after a successful CN negotiation in the SIP SDP. If the
Silence Suppression
client endpoint's OPUS codec supports the reception of DTX packets, the UCM will send DTX packets instead.

SRTP Enable SRTP for the VoIP trunk. The default setting is "No".

SRTP Crypto Suite SRTP encryption suite used by UCM for outbound calls. Priority is based on order of configuration.

Similar to Enable Direct Media. The PBX will attempt to redirect the RTP media streams to bypass the PBX and to go directly
IPVT Mode
between caller and callee. Primarily for use with trunks to IPVT.

Enable T.38 UDPTL Enable or disable T.38 UDPTL support.

Carry the ssrc/msid/mid/ct/as/tias/record properties of the SDP. These attributes may cause incompatibility when
Special attributes
interconnecting with other devices.

Send PPI Header If checked, the invite message sent to trunks will contain PPI (P-Preferred-Identity) Header.

If checked, the INVITE, 18x and 200 SIP messages sent to trunks will contain P-Asserted-Identity (PAI) header. It is not
Send PAI Header
possible to send both PPI and PAI headers.

DOD as From Name If enabled and "From User" is configured, the INVITE's From header will contain the DOD number.

Passthrough PAI
If enabled and "Send PAI Header" is disabled, PAI headers will be preserved as calls pass through the UCM.
Header

Send PANI Header If checked, the INVITE and REGISTER sent to the trunk will contain P-Access-Network-Info header.

Send Anonymous If checked, the "From" header in outgoing INVITE message will be set to anonymous.

Outbound Proxy
Enable to send outbound signal to the proxy instead of the devices directly.
Support

DID Mode Configure to obtain the destination ID of an incoming SIP call from SIP Request-line or To header.

GIN Registration If enabled, the UCM will send a GIN REGISTER (generate implicit numbers).

Configure the mode for sending the DTMF.

● RFC4833 (default): DTMF is transmitted as audio in the RTP stream but is encoded separately from the audio stream.
Backwards-compatible with RFC2833.

● DTMF is transmitted as audio and is included in the audio stream. Requires alaw/ulaw codecs
DTMF Mode
● Info: DTMF is transmitted separately from the media streams.

● RFC4733_info: DTMF is transmitted through both RFC4733 and SIP INFO

● Auto: DTMF mode will be negotiated with the remote peer, only supports RFC4733 and inband. RFC4733 will be used by
default unless the remote peer does not indicate support.
 Enable Heartbeat
If enabled, the PBX will regularly send SIP OPTIONS to check if the device is online.
detection

The Maximum
The number of current outgoing calls over the trunk at the same time. The default value 0 means no limit.
Number of Call Lines

Automatically sync local LDAP phonebooks to a remote peer (SIP peer trunk only). To ensure successful syncing, the remote

peer must also enable this service and set the same password as the local UCM. Port 873 is used by default.

● Sync LDAP Password: Password used for LDAP phonebook encryption and decryption. The password must be the same
for both peers to ensure successful syncing.

● LDAP Outbound Rule: Specify an outbound rule. The PBX system will automatically modify the remote contacts by
Sync LDAP Enable
adding prefix parsed from this rule.

● LDAP Dialed Prefix: System will automatically modify the remote contacts by adding this prefix

● LDAP Sync Method: Specifies the sync method of the UCM. When an LDAP sync request is received, the UCM will send
the phonebook data via the specified method.

● LDAP Last Sync Date: The last successful sync date.

Block Spam Calls.

● Disabled: Disable STIR/SHAKEN.

STIR/SHAKEN ● Outgoing Attest: Enable STIR/SHAKEN attestation for outgoing calls.

● Incoming Verify: Enable STIR/SHAKEN verification for incoming calls.

● Both: Enable STIR/SHAKEN for both outgoing and incoming calls

Check this box to allow the system to automatically alert this extension when a called party is available, given that a previous

Enable CC call to that party failed for some reason. If the Call Waiting is disabled, the CC service will take effect only for unanswered and

timeout calls.

Select the VoIP trunk type.

Type ● Peer IAX Trunk

● Register IAX Trunk

Provider Name Configure a unique label to identify this trunk when listed in outbound rules, inbound rules, etc.

Host Name Configure the IP address or URL for the VoIP provider’s server of the trunk.

If enabled, the trunk CID will not be overridden by the extension's CID when the extension has CID configured. The default setting is
Keep Trunk CID
"No".

Username Enter the username to register to the trunk from the provider when the "Register IAX Trunk" type is selected.

Password Enter the password to register to the trunk from the provider when the "Register IAX Trunk" type is selected.

Disable This Trunk If selected, the trunk will be disabled.

 The number that the trunk will try to use when making outbound calls.

CID priority from highest to lowest is as follows:

From User (register trunk only) >>> Inbound Call CID (if Keep Original CID is enabled and the call is originally from another trunk)

Caller ID Number >>> Trunk CID (Keep Trunk CID enabled) >>> DOD CID >>> Extension CID >>> Register Trunk Username (Keep Trunk CID

disabled) >>> Global Outbound CID.

Note 1: Certain providers may ignore this CID.

Note 2: If this CID contains an asterisk (*), call recordings from this trunk might be lost when saving them to NAS storage.

CallerID Name Configure the new name of the caller when the extension has no CallerID Name configured.

Table 63: Create New IAX Trunk

IAX Peer Trunk IAX Register Trunk

Basic Settings

Provider Name Configure a unique label to identify this trunk when listed in outbound rules, inbound rules, etc.

Host Name Configure the IP address or URL for the VoIP provider’s server of the trunk.

If enabled, the trunk CID will not be overridden by the extension’s CID when the extension has CID configured. The default
Keep Trunk CID
setting is “No”.

Disable This Trunk If selected, the trunk will be disabled.

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it

might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will

Caller ID be set before sending out the call.

From the user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID

(Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled)

🡪 Global Outbound CID.

CallerID Name Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Username Enter the username to register to the trunk from the provider.

Password Enter the password to register to the trunk from the provider.

Advanced Settings

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726,
Codec Preference
G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8.

Enable Heartbeat If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default

Detection setting is “No”.

 Heartbeat When the “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message

Frequency sent to the device to check if the device is still online. The default setting is 60 seconds.

Maximum Number
The maximum number of concurrent calls using the trunk. The default setting is 0, which means no limit.
of Call Lines

Trunk Groups
Users can create VoIP Trunk Groups to register and easily apply the same settings on multiple accounts within the same SIP server. This can drastically reduce

the amount of time needed to manage accounts for the same server and improve the overall cleanliness of the web UI.

Trunk Group

Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the configured SIP server by pressing the

button and configuring the username, password, and authentication ID fields.

Trunk Group Configuration

Type Register Trunk

Provider Name Configure a unique label to identify the trunk when listed in outbound rules and incoming rules.

Host Name Enter the IP address or hostname of the VoIP provider's server.

Transport Configure the SIP Transport method. Using TCP requires local TCP support; using TLS requires local TLS support.

Keep CID from the inbound call when dialing out even if option "Keep Trunk CID" is enabled. Please make sure the peer
Keep Original CID
PBX at the other end supports matching user entry using the "username" field from the authentication line.

Keep Trunk CID Always use trunk CID if specified even if extension has DOD number or CID configured.

Enable this setting if the UCM is using public IP and communicating with devices behind NAT.

NAT Note 1: This setting will overwrite the Contact header of received messages, which may affect the ability to establish calls

when behind NAT. Consider changing settings in PBX Settings 🡪 SIP Settings 🡪 NAT instead.

Disable This Trunk Check this box to disable this trunk

if "Enabled" option is selected, TEL URI and remove OBP from Route cannot be enabled at the same time. If the phone has

an assigned PSTN telephone number, this field should be set to "User=Phone". A "User=Phone" parameter will the be
TEL URI
attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will

be used instead of "SIP:" in the SIP request.

Need Registration Whether to register on the external server.

Allow outgoing calls if

Uncheck to block outgoing cakks if registration fails. If "Need Registration" option is unchecked, this settting will be ignored.
registration fails

CallerID Name To display the caller ID name of the trunk, you must configure the caller ID number of the trunk.

The number used to register with the provider server, and the VoIP provider will authenticate the number based on the trunk
Trunk Registration Number
registration number.

Linear: Select lines in list order and make Outbound calls. Round Robin: Rotary line selection with memory and making
Line Selection Strategy
Outboun calls.

AuthTrunk If enabled, the UCM will send a 401 response to the incming call to authenticate the trunk.

Auto Record If enabled, calls handled with this extension/trunk will automatically be recorded.

Direct Callback Allows external numbers the option to get directed to the extension that last called them.

If enabled, RemoteConnect-related options will be automatically configured. Please confirm the trunk has a GDMS-assigned
RemoteConnect Mode
address or supports TLS.

If enabled, the number of concurrent calls on this trunk will be monitored. If the "Trunk Concurrent Calls" system alert is

Monitor Concurrent Calls enabled, alert notifications will be generated if the number of concurrent calls exceeds this trunk's configured concurrent call

thresholds.
 Concurrent Call Threshold Threshold of all incoming and outgoing concurrent calls in this trunk.

Outgoing Concurrent Call

Threshold of all outgoing concurrent calls passing through this trunk.
Threshold

Incoming Concurrent Call

Threshold of all incoming concurrent calls passing through this trunk.
Threshold

Enable Total Time Limit For If enabled, a limit will be placed on the cumulative duration of outbound calls within a specific period. Once this limit has

Outbound Calls been reached, further outbound calls from this trunk will not be allowed.

Direct Outward Dialing (DOD)

The UCM630X provides Direct Outward Dialing (DOD), which is a service of a local phone company (or local exchange carrier) that allows subscribers

within a company’s PBX system to connect to outside lines directly.

Example of how DOD is used:

Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated with it. The main number of the office is routed to an auto attendant. The other three

numbers are direct lines to specific users of the company. Now when a user makes an outbound call their caller ID shows up as the main office number. This

poses a problem, as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD for the CEO’s extension.

Steps to configure DOD on the UCM630X:

1. To setup DOD go to UCM630X Web GUI🡪Extension/Trunk🡪VoIP Trunks page.

2. Click to access the DOD options for the selected SIP Trunk.

3. Click “Add DOD” to begin your DOD setup

4. Enter a SIP trunk DID number in the “DOD number” field. In this example, ABC company has a total of 4 DID numbers. Enter the phone number used

by the CEO here.

5. When adding extensions, you can choose whether to “Enable Strip” according to your needs. If it is enabled, you can configure the number (0-64) that
will be stripped from the extension number before being added to the DOD number. For example, if the entered digit is 2, and the DOD number for

extension 4002 is 1122, then dialing out from 4002, 112202 will be used as the caller ID (DOD).

6. Select an extension from the “Available Extensions” list. Users have the option of selecting more than one extension. In this case, Company ABC would
select the CEO’s extension. After making the selection, click on the button to move the extension(s) to the “Selected Extensions” list.
 DOD extension selection

7. Click “Save” at the bottom.

Once completed, the user will return to the EDIT DOD page that shows all the extensions that are associated with a particular DOD.

Edit DOD

: Add a DOD.

: Import DODs using a csv file.

: Export the DODs using a csv file.

: Filter DODs by number or name.

For DOD importing, please refer to the screenshot below for the template used.

DOD CSV file Template

WebRTC Trunks
WebRTC, Web Real-Time Communication, is a real time audio/video chatting framework that allows real-time audio/video chatting through the web browser.

WebRTC usually does not refer to the web application itself but to the set of protocols and practices bundled with a graphical interface. Our UCM63xx
supports creating WebRTC trunks to use exclusively with web application, this allows the users to join calls and meetings just by clicking a link to a web

page.

Below is a figure that shows the options to configure when setting up this feature:

Trunk Name Create a unique label to easily identify the trunk for inbound route configuration.

Disable This Trunk Check this box to disable this trunk.

Auto Record If enabled, calls handled with this extension/trunk will automatically be recorded.

Select jitter buffer method for temporary accounts such as meeting participants who joined via link.

Disable: Jitter buffer will not be used.

Jitter Buffer Fixed: Jitter buffer with a fixed size (equal to the value of "Jitter Buffer Size")

Adaptive: Jitter buffer with a adaptive size that will not exceed the value of "Max Jitter Buffer").

NetEQ: Dynamic jitter buffer via NetEQ.

If enabled, the number of concurrent calls on this trunk will be monitored. If the "Trunk Concurrent Calls" system alert is

Monitor Concurrent Calls enabled, alert notifications will be generated if the number of concurrent calls exceeds this trunk's configured concurrent call

thresholds.

Incoming Concurrent Call

Threshold of all incoming concurrent calls passing through this trunk.
Threshold
 WebRTC Inbound Link This link can be embedded onto a web page. Clicking the link will connect to a pre-configured WebRTC trunk destination. You

Address can also enter this link in the browser address bar to directly access and test WebRTC calls.

Important Note

Please note that in order to use WebRTC Trunk feature, the UCM must have a paid RemoteConnect plan. For more information, please refer to the following link:
https://ucmrc.gdms.cloud/plans

SLA STATION
The UCM630X supports SLA (Shared Line Appearance) that allows mapping the key with LED on a multi-line phone to different external lines. When there

is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to

know if the line is occupied or not. The SLA function on the UCM630X is like BLF but SLA is used to monitor external lines, i.e., analog trunk on the

UCM630X. Users could configure the phone with BLF mode on the MPK to monitor the analog trunk status or press the line key pick-up call from the analog

trunk on the UCM630X.

Create/Edit Shared Line Appearance Station

Shared Line Appearance Station can be configured on Web GUI🡪Extension/Trunk🡪SLA Station.

SLA Station

Click on to add an Shared Line Appearance Station.

Click on to edit the Shared Line Appearance Station. The following table shows the SLA Station configuration parameters.

Click on to delete the Shared Line Appearance Station.

Station
Configure a name to identify the SLA Station.
Name

Station Specify a SIP extension as a station that will be using SLA.

Available
Existing Analog Trunks with SLA Mode enabled will be listed here.
SLA Trunks

Select a trunk for this SLA from the Available SLA Trunks list. Click on to arrange the order. If there are multiple trunks
Associated
selected, when there are calls on those trunks at the same time, pressing the LINE key on the phone will pick up the call on the first trunk
SLA Trunks
here.

SLA Station

Options
Ring Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there
Timeout will be no timeout.

Configure the time (in seconds) for a delay before ringing the station when a call first comes in on the shared line. No delay is set by
Ring Delay
default. If set to 0, there will be no delay.

This option defines the competence of the hold action for one particular trunk. If set to “open”, any station could hold a call on that trunk or

Hold Access resume one held session; if set to “private”, only the station that places the trunk call on hold could resume the session. The default setting

is “open”.

Table 66: SLA Station Configuration Parameters

Sample Configuration
1. On the UCM630X, go to Web GUI🡪Extension/Trunk🡪Analog Trunks page. Create an analog trunk or edit the existing analog trunk. Make sure “SLA
Mode” is enabled for the analog trunk. Once enabled, this analog trunk will be only available for the SLA stations created under Web

GUI🡪Extension/Trunk🡪SLA Station page.

Enable SLA Mode for Analog Trunk

2. Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.

Analog Trunk with SLA Mode Enabled

3. On the UCM630X, go to Web GUI🡪Extension/Trunk🡪SLA Station page, click on “Add”. Please refer to section [Create/Edit SLA Station] for the

configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows station 1005 associated
with FXO1 trunk.
 SLA Example – SLA Station

SLA Example -SLA Created

4. On the SIP phone 2, configure to register UCM630X extension 1005. Configure the MPK as BLF mode and the value must be set to

“extension_trunkname”, which is 1005_fxo1 in this case.

SLA Example – MPK Configuration

Now the SLA station is ready to use. The following functions can be achieved by this configuration.

Making an outbound call from the station/extension, using LINE key

When the extension is in an idle state, press the line key for this extension on the phone to off-hook. Then dial the station’s extension number, for example,
dial 1002 on phone 1 (or dial 1005 on phone 2), to hear the dial tone. Then the users could dial an external number for the outbound call.

Making an outbound call from the station/extension, using the BLF key

When the extension is in an idle state (BLF is lit in green), press the MPK and users could dial external numbers directly.

Answering call using LINE key

When the station is ringing (BLF is blinking in red), press the LINE key to answer the incoming call.

Barging-in active call using BLF key

When there is an active call between an SLA station and an external number using the SLA trunk, other SLA stations monitoring the same trunk could join the

call by pressing the BLF key if “Barge Allowed” is enabled for the analog trunk.

Hold/UnHold using BLF key

If the external line is previously put on hold by an SLA station, another station that monitors the same SLA trunk could UnHold the call by pressing the BLF

key if “Hold Access” is set to “open” on the analog trunk and the SLA station.

CALL ROUTES

Outbound Routes
In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in UCM630X, these rules are the regulating

points for all external outgoing calls initiated by the UCM through all types of trunks: SIP, Analog, and Digital.

Configuring Outbound Routes

In the UCM630X, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through

different trunks (e.g., “Local” 7-digit dials through an FXO while “Long distance” 10-digit dials through a low-cost SIP trunk). Users can also set up a

failover trunk to be used when the primary trunk fails.

Go to Web GUI🡪Extension/Trunk🡪Outbound Routes to add and edit outbound rules.

Click on to add a new outbound route.

Click the “Import” button to upload the outgoing route in .CSV format.

Click the “Export” button to generate outgoing routes in .CSV format.

Click to edit the outbound route.

Click to delete the outbound route.

On the UCM630X, the outbound route priority is based on the “Best matching pattern”. For example, the UCM630X has outbound route A with pattern 1xxx

and outbound route B with pattern 10xx configured. When dialing 1000 for an outbound call, outbound route B will always be used first. This is because

pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.

Outbound Rule
Configure the name of the calling rule (e.g., local, long_distance, etc.). Letters, digits, _ and – are allowed.
Name
 All patterns are prefixed by the “_” character, but please do not enter more than one “_” at the beginning. All patterns can add
comments, such as “_pattern /* comment */”. In patterns, some characters have special meanings:

[12345-9] … Any digit in the brackets. In this example, 1,2,3,4,5,6,7,8,9 is allowed.

N … Any digit from 2-9.

. … Wildcard, matching one or more characters.

Pattern
! … Wildcard, matching zero or more characters immediately.

X … Any digit from 0-9.

Z … Any digit from 1-9.

– … Hyphen is to connect characters and it will be ignored.

[] Contain special characters ([x], [n], [z]) represent letters x, n, z.

Disable This After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore.

Route However, the route settings will remain in UCM. Users can enable it again when it is needed.

Password Configure the password for users to use this rule when making outbound calls.

Local Country
If your local country code is affected by the outbound blacklist, please enter it here to bypass the blacklist.
Code

Call Duration
Enable to configure the maximum duration for the call using this outbound route.
Limit

Maximum Call
Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limit.
Duration

Configure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when
Warning Time
x seconds are left to end the call.

Auto Record If enabled, calls using this route will automatically be recorded.

Warning
Configure the warning repeat interval for the call using this outbound route. If set to X seconds, the warning tone will be played every x
Repeat
seconds after the first warning.
Interval

PIN Groups Select a PIN Group

PIN Groups

with Privilege If enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied.

Level
 Select privilege level for the outbound rule.

Internal: The lowest level required. All users can use this rule.

Local: Users with Local, National, or International levels can use this rule.

National: Users with National or International levels can use this rule.
Privilege Level
International: The highest level required. Only users with the international level can use this rule.

Disable: The default setting is “Disable”. If selected, only the matched source caller ID will be allowed to use this outbound route.

Please be aware of the potential security risks when using the “Internal” level, which means all users can use this outbound rule to dial
out from the trunk.

When enabled, users could specify extensions allowed to use this outbound route. “Privilege Level” is automatically disabled if using
“Enable Filter on Source Caller ID”.

The following two methods can be used at the same time to define the extensions as the source caller ID.

1. Select available extensions/extension groups from the list. This allows users to specify arbitrary single extensions available in the

PBX.

2. Custom Dynamic Route: define the pattern for the source caller ID. This allows users to define extension range instead of selecting
them one by one.

All patterns are prefixed with the “_”.

Enable Filter
Special characters:
on Source
Caller ID X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

“.“: Wildcard. Match one or more characters.

“!“: Wildcard. Match zero or more characters immediately.

Example: [12345–9] – Any digit from 1 to 9.

Note: Multiple patterns can be used. Patterns should be separated by a comma “,”. Example: _X. , _NNXXNXXXXX, _818X.

Outbound
Attempt to use the configured outbound route CID. This CID will not be used if DOD is configured.
Route CID

Send This Call

Through

Trunk

Trunk Select the trunk for this outbound rule.

 Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via
the selected trunk.

Strip Example:

The users will dial 9 as the first digit of long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In
this case, 1 digit should be stripped before the call is placed.

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is
Prepend
stripped.

Use Failover

Trunk

Failover trunks can be used to make sure that a call goes through an alternate route when the primary trunk is busy or down. If “Use
Failover Trunk” is enabled and “Failover trunk” is defined, the calls that cannot be placed via the regular trunk may have a secondary

trunk to go through.

UCM630X supports up to 10 failover trunks.

Failover Trunk

Example:

The user’s primary trunk is a VoIP trunk, and the user would like to use the PSTN when the VoIP trunk is not available. The PSTN trunk

can be configured as the failover trunk of the VoIP trunk.

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via

the selected trunk.

Strip Example:

The users will dial 9 as the first digit of long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In

this case, 1 digit should be stripped before the call is placed.

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is
Prepend
stripped.

Time
Condition

Use Main Trunk or Failover Trunk: Use the Main Trunk and its settings during the configured time conditions. If the main trunk is
Time unavailable, the Failover Trunk and its settings will be used instead.
Condition
Mode Use Specific Trunks: Use specific trunks during the configured time conditions. The Strip and Prepend settings of the Main Trunk will

be used. If a trunk is unavailable during its time condition, no failover trunks will be used.

Table 67: Outbound Route Configuration Parameters

Failover Trunk Toggles

Inbound Routes

This option controls whether failover trunks will be used if receiving specific responses to outgoing calls.

Failover Trunk Toggles

If a call receives the selected response codes, the UCM will not be redirect it to the call route’s failover trunk.

Note

Due to the addition of this option, the Enable 486 to Failover Trunks option under PBX Settings🡪General Settings page has been removed.

Outbound Routes DOD

It is possible to specify DOD number based on Outbound Route, as displayed on the screenshot below. For each outbound route.

Outbound Routes Page

 DOD Configuration by Outbound Route

Outbound Blacklist

The UCM630X allows users to configure a blacklist for outbound routes. If the dialing number matches the blacklist numbers or patterns, the outbound call

will not be allowed. The outbound blacklist can be configured under UCM Web GUI🡪Extension/Trunk🡪Outbound Routes: Outbound Blacklist.

Users can configure numbers, patterns or select country code to add to the blacklist. Please note that the blacklist settings apply to all outbound routes.

Country Codes

Users can export outbound route blacklists and delete all blacklist entries. Additionally, users can also import blacklists for outbound routes.
 Blacklist Import/Export

Scheduled Sync

The UCM630X allows users to synchronize the outbound routes, this feature can be found on the Web GUI🡪Extension/Trunk🡪Outbound Routes🡪
Scheduled Sync.

Scheduled Sync Enable the Scheduled Sync feature

Server Address Enter the TFTP server address. For example, “192.168.1.2:69”.

File Name Specify the file name

Sync Time Enter the sync time (24hr format). The valid range is 0-23.

Sync Frequency Create new sync every x day(s). The valid range is 1 to 30.

Table 68: Outbound Routes/Scheduled Sync

PIN Groups

The UCM630X supports the pin group. Once this feature is configured, users can apply pin groups to specific outbound routes. When placing a call on pin-
protected outbound routes, the caller will be asked to input the group pin number, this feature can be found on the Web GUI🡪Extension/Trunk🡪Outbound

Routes🡪PIN Groups.

Name Specify the name of the group

Record In CDR Specify whether to enable/disable record in CDR

PIN Number Specify the code that will be asked once dialing via a trunk

PIN Name Specify the name of the PIN

Table 69: Outbound Routes/PIN Group

Once the user clicks , the following figure shows to configure the new PIN.
 Create New PIN Group

The following screenshot shows an example of created PIN Groups and members:

PIN Members

If PIN group is enabled on outbound route level, password, privilege level and enable the filter on source caller ID will be disabled, unless if you check the

option “PIN Groups with Privilege Level” where you can use the PIN Groups and Privilege Level or PIN Groups and Enable Filter on Source Caller ID.

Outbound PIN

If PIN group CDR is enabled, the call with PIN group information will be displayed as part of CDR under the Account Code field.

CDR Record

Importing PIN Groups from CSV files:

Users can also import PIN Groups by uploading CSV files for each group. To do this:

1. Navigate to Extension/Trunk🡪Outbound Routes🡪PIN Groups and click on the “Upload” button.

Importing PIN Groups from CSV files

2. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:

Incorrect CSV File

3. To ensure a successful import, please follow the format in the sample image below

CSV File Format

The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.

Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not be changed and will cause an upload error otherwise.

Rows 3+ contain the user-defined values with Column A holding the PINs and Column B holding the PIN names. PIN values must consist of at least four

digits.

Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.

CSV File Successful Upload

Inbound Routes
Inbound routes can be configured via Web GUI🡪Extension/Trunk🡪Inbound Routes.

Click on to add a new inbound route.

Click on “Blacklist” to configure the blacklist for all inbound routes.

Click on to edit the inbound route.

Click on to delete the inbound route.

Inbound Rule Configurations

Trunks Select the trunk to configure the inbound rule.

Inbound Route Name Configure the name of the Inbound Route. For example, “Local”, “LongDistance” etc.

All patterns are prefixed with the “_”.

Special characters:

X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. “.“: Wildcard. Match one or more characters. “!“: Wildcard.

Match zero or more characters immediately. Example: [12345-9] – Any digit from 1 to 9.

Pattern Notes:

Multiple patterns can be used. Each pattern should be entered in a new line.

Example:

_X.

_ NNXXNXXXXX /* 10-digit long distance */

After creating the inbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore.
Disable This Route
However, the route settings will remain in UCM. Users can enable it again when it is needed.

Configures the source of the CID to match with the configured CallerID Pattern.

None: CID is not obtained from any source. Only applicable if no CallerID Pattern is configured.

CID Source DiversionID: CID is obtained from the Diversion header. Only applicable to SIP trunks.

CallerID: If the call is from a SIP trunk, the CID is obtained from the From header. Otherwise, the CID will be obtained from other

related signaling.

Seamless Transfer Allows the selected extension to use this function. If an extension is busy, and a mobile phone is bound to that extension, the mobile

Whitelist phone can pick up calls to that extension.

Ringback tone Choose the custom ringback tone to play when the caller reaches the route.

Auto Record If enabled, calls using this route will automatically be recorded.

If enabled, collect calls will be blocked.

Block Collect Call Note: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-

Info: service-code=Collect Call”.

 Configure the Alert-Info, when UCM receives an INVITE request, the Alert-Info header field specifies an alternative ring tone to
Alert-Info
the UAS.

Fax Detection If enabled, fax signals from the trunk during a call will be detected.

Configures the destination of faxes.

● Extension: send the fax to the designated FXS/SIP extension (fax machine) or a FAX extension.

● Fax to Email: send the fax as an email attachment to the designated extension’s email address. If the selected extension does not
Fax Destination
have an associated email address, it will be sent to the default email address configured in the Call Features->Fax/T.38->Fax

Settings page.

Note: please make sure the sending email address is correctly configured in System Settings->Email Settings.

Prepend Trunk Name If enabled, the trunk name will be added to the caller id name as the displayed caller id name.

Manipulates Caller ID (CID) name and/or number within the call flow to help identify who is calling. When enabled two fields will
Set Caller ID Info
show allowing to manipulate the CalleID Number and the Caller ID Name.

Configure the pattern-matching format to manipulate the numbers of incoming callers or to set a fixed CallerID number for calls

that go through this inbound route.

● ${CALLERID(num)}: Default value which indicates the number of an incoming caller (CID). The CID will not be modified.

● ${CALLERID(num):n}: Skips the first n characters of a CID number, where n is a number.

CallerID Number
● ${CALLERID(num):-n}: Takes the last n characters of a CID number, where n is a number.

● ${CALLERID(num):s:n}: Takes n characters of a CID number starting from s+1, where n is a number and s is a character
position (e.g. ${CALLERID(num):2:7} takes 7 characters after the second character of a CID number).

● n${CALLERID(num)}: Prepends n to a CID number, where n is a number.

The default string is ${CALLERID(name)}，which means the name of an incoming caller, it is a pattern-matching syntax format.

CallerID Name A${CALLERID(name)}B means Prepend a character ‘A’ and suffix a character ‘B’ to ${CALLERID(name)}.

Not using pattern-matching syntax means setting a fixed name to the incoming caller.

Gives uses the ability to configure inbound mode per individual route. When enabled two fields will show allowing to set the
Enable Route-Level
Inbound mode and the Inbound mode Suffix.
Inbound Mode
Note: Global inbound mode must be enabled before users can configure route-level inbound mode.

Choose the inbound mode for this route.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the
Inbound Mode
option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be

affected by the global inbound mode change.

Dial “Global Inbound Mode feature code + Inbound Mode Suffix” or a route’s assigned suffix to toggle the route’s inbound mode.
Inbound Mode Suffix
The BLF subscribed to the inbound mode suffix can monitor the current inbound mode.

Multiple mode allows users to switch between destinations of the inbound rule by feature codes. Configure related feature codes as
Inbound Multiple
described in [Inbound Route: Multiple Mode]. If this option is enabled, the user can use feature code to switch between different
Mode
modes/destinations.
 If enabled, the callerID will be resolved to a name through local LDAP. Note, if a matched name is found, the original callerID

CallerID Name Lookup name will be replaced. The name lookup is performed before other callerID or callerID name modifiers (e.g., Inbound Route's Set

CallerID Info or Prepend Trunk Name). Note: Name lookup may impact system performance.

This option shows up only when “By DID” is selected. If enabled, the external users dialing into the trunk via this inbound route
Dial Trunk
can dial outbound calls using the UCM’s trunk.

This option shows up only when “By DID” is selected.

● Disable: Only the selected Extensions or Extension Groups are allowed to use this rule when enabled Filter on Source Caller ID.

● Internal: The lowest level required. All users are allowed to use this rule, checking this level might be risky for security
Privilege Level purposes.

● Local: Users with Local level, National or International level are allowed to use this rule.

● National: Users with National or International Level are allowed to use this rule.

● International: The highest level required. Only users with an international level are allowed to use this rule.

This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via

the inbound route. The DID destination is:

● Extension

● Conference

● Call Queue
Allowed DID
● Ring Group
Destination
● Paging/Intercom Groups

● IVR

● Voicemail Groups

● Dial By Name

● All

Select the default destination for the inbound call.

● Extension

● Voicemail

● Conference Room

● Call Queue

● Ring Group

● Paging/Intercom

● Voicemail Group

Default Destination ● DISA

● IVR

● External Number

● By DID

When “By DID” is used, the UCM will look for the destination based on the number dialed, which could be local extensions,

conference, call queue, ring group, paging/intercom group, IVR, and voicemail groups as configured in “DID destination”. If the

dialed number matches the DID pattern, the call will be allowed to go through.

● Dial By Name

● Callback
 Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is
Strip
stripped.

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is
Prepend
stripped.

Time Condition

Start Time Select the start time “hour:minute” for the trunk to use the inbound rule.

End Time Select the end time “hour:minute” for the trunk to use the inbound rule.

Date Select “By Week” or “By Day” and specify the date for the trunk to use the inbound rule.

Week Select the day in the week to use the inbound rule.

Select the destination for the inbound call under the defined time condition.

● Extension

● Voicemail

● Conference Room

● Call Queue

● Ring Group

● Paging/Intercom

● Voicemail Group

● DISA
Destination ● IVR

● By DID

When “By DID” is used, the UCM will look for the destination based on the number dialed, which could be local extensions,

conference, call queue, ring group, paging/intercom group, IVR, and voicemail groups as configured in “DID destination”. If the

dialed number matches the DID pattern, the call will be allowed to go through.

Configure the number of digits to be stripped in the “Strip” option.

● Dial By Name

● External Number

● Callback

Inbound Route: Prepend Example

UCM630X allows users to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in the

inbound route DID pattern, the user no longer needs to create multiple routes for the same trunk to route calls to different extensions. The following example

demonstrates the process:

1. If Trunk provides a DID pattern of 18005251163.

2. If Strip is set to 8, UCM630X will strip the first 8 digits.

3. If Prepend is set to 2, UCM630X will then prepend a 2 to the stripped number, now the number becomes 2163.

4. UCM630X will now forward the incoming call to extension 2163.

 Inbound Route feature: Prepend

Inbound Route: Multiple Mode

In the UCM630X, the user can configure an inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can

be enabled under Inbound Route settings.

Inbound Route – Multiple Mode

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode 1” destination for all routes. By default,

the call coming into the inbound routes will be routed to the default destination.

SIP end devices that have registered on the UCM630X can dial feature code *62 to switch to inbound route “Mode 1” and dial feature code *61 to switch

back to “Default Destination”. Switching between different mode can be easily done without Web GUI login.
For example, the customer service hotline destination has to be set to a different IVR after 7 PM. The user can dial *62 to switch to “Mode 1” with that IVR

set as the destination before off work.

To customize feature codes for “Default Mode” and “Mode 1”, click on under the “Inbound Routes” page, check the “Enable

Inbound Multiple Mode” option, and change “Inbound Default Mode” and “Inbound Mode 1” values (By default, *61 and *62 respectively).

Inbound Route – Multiple Mode Feature Codes

Inbound Route: Route-Level Mode

In the UCM630X, users can enable Route-Level Inbound Mode to switch between different destinations for each inbound route. The inbound Route-Level

mode can be enabled under Inbound Route settings.

Inbound Route – Route-Level Mode

The global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, Mode 1 must be configured as well.

When Route-Level Inbound Mode is enabled, the user can configure a “Default Destination” and a “Mode 1” destination for each specific route. By default,
the call coming into this specific inbound route will be routed to the default destination.

Users can toggle the route’s inbound mode by dialing “Global Inbound Mode feature code + Inbound Mode Suffix” and the current inbound route can be

monitored by subscribing a BLF to the Inbound Mode Suffix.

For example, the Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to 1010. To switch the mode of route 1 to

Default Mode, users can dial *611010.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the

global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.

Inbound Route: Inbound Mode BLF Monitoring

Users can assign MPKs and VPKs to monitor and toggle the current global inbound mode of the UCM.

To do this, please refer to the following steps:

1. Access the UCM web GUI and navigate to Extension/Trunk🡪Inbound Routes.

2. Click on the button and enable Inbound Multiple Mode.

3. Edit the subscribe number field to the desired BLF value.

Global Inbound Mode

4. Configure the BLF value on a phone’s MPK/VPK. As an example, a GXP2140 with the BLF configured will show the Inbound Mode status on its screen

once configured. The 777 BLF is lit green, indicating that the current inbound mode is “Default Mode”.

Inbound Mode – Default Mode

5. Pressing the key will toggle the inbound mode to “Mode 1”, and the button’s color will change to red.
 Inbound Mode – Mode 1

Inbound Route: Import/Export Inbound Route

Users can now import and export inbound routes to quickly set up inbound routing on a UCM or to back up an existing configuration. An exported inbound

route configuration can be directly imported without needing any manual modifications.

Import/Export Inbound Route

The imported file should be in CSV format and using UTF-8 encoding, the imported file should contain below columns, and each column should be separated

by a comma (It is recommended to use Notepad++ for the imported file creation):

Disable This Route: Yes/No.

Pattern: Always prefixed with _

CallerID Pattern: Always prefixed with _

Prepend Trunk Name: Yes/No.

Prepend User Defined Name Enable: Yes/No.

Prepend User Defined Name: A string.

Alert-info: None, Ring 1, Ring 2… User should enter an Alert-info string following the values we have in the Inbound route Alert-Info list.

Allowed to seamless transfer: [Extension_number]

Inbound Multiple Mode: Yes/No.

Default Destination: By DID, Extension, Voicemail… Users should enter a Default Destination string following the values we have in the Inbound route
Default Destination list.

Destination: An Extension number, Ring Group Extension…

Default Time Condition.

Mode 1: By DID, Extension, Voicemail… Users should enter a Default Destination string following the values we have in the mode 1 Default

Destination list.

Mode 1 Destination: An Extension number, Ring Group Extension…

Mode 1 Time Condition.

FAX with Two Media

The UCM630X supports Fax re-INVITE with multiple codec negotiation. If a Fax re-INVITE contains both T.38 and PCMA/PCMU codec, UCM630X will

choose T.38 codec over PCMA/PCMU.

Blacklist Configurations

In the UCM630X, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on “Blacklist”.
 Select the checkbox for “Blacklist Enable” to turn on the Blacklist feature for all inbound routes. Blacklist is disabled by default.

Enter a number in the “Add Blacklist Number” field and then click ”Add” to add to the list. Anonymous can also be added as a Blacklist Number by
typing “Anonymous” in Add Blacklist Number field.

To remove a number from the Blacklist, select the number in the “Blacklist list” and click on or click on the” Clear” button to remove all the numbers
on the blacklist.

Users can also export the inbound route blacklist by pressing the button.

Blacklist Configuration Parameters

To add blacklisted numbers in batch, click on “Import” to upload the blacklist file in CSV format. The supported CSV format is as below.

Blacklist CSV File

Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for “Blacklist Add’ (default: *40) and “Blacklist

Remove” (default: *41) from an extension. The feature code can be configured under Web GUI🡪Call Features🡪Feature Codes.

FAX SERVER
The UCM6300 series supports T.30/T.38 Fax and Fax Pass-through. It can convert the received Fax to PDF format and send it to the configured Email

address. Fax/T.38 settings can be accessed via Web GUI🡪Call Features🡪FAX/T.38. The list of received Fax files will be displayed on the same web page
for users to view, retrieve and delete.

Configure Fax/T.38
Click on “Create New Fax Extension”. In the popped-up window, fill the extension, name, and Email address to send the received Fax to.
 Click on “Fax Settings” to configure the Fax parameters.

Click on to edit the Fax extension.

Click on to delete the Fax extension.

Fax Settings

Enable Error Configure to enable Error Correction Mode (ECM) for the Fax.
Correction

Mode The default setting is “Yes”.

Configure the maximum transfer rate during the Fax rate negotiation.

Maximum
The possible values are 2400, 4800, 7200, 9600, 12000, and 14400.
Transfer Rate

The default setting is 14400.

Minimum Configure the minimum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000, and
Transfer Rate 14000. The default setting is 2400.

Configure the concurrent fax that can be sent by UCM6300. Two modes “Only” and “More” are supported.

Only

Max Under this mode, the UCM6300 allows only a single user to send a fax at a time.
Concurrent

Sending Fax More

Under this mode, the UCM6300 supports multiple concurrent faxes sending by the users.

By default, this option is set to “only”.

Configure the maximum length of Fax Queue from 6 to 10.

Fax Queue

Length
The default setting is 6.
 User
Information in If enabled, this will give users the option to send a special header in SIP fax messages.

Fax Header

Fax Header
Adds fax header into the fax file.
Information

Configure the Email address to send the received Fax to if the user’s Email address cannot be found.

Default Email Note:

Address
The extension’s Email address or the Fax’s default Email address needs to be configured to receive Fax from Email. If neither of them

is configured, Fax will not be received from email.

Fill in the “Subject:” and “Message:” content, to be used in the Email when sending the Fax to the users. The template variables are:

${CALLERIDNUM} : Caller ID Number

${CALLERIDNAME} : Caller ID Name

Template
${RECEIVEEXTEN} : The extension to receive the Fax
Variables

${FAXPAGES} : Number of pages in the Fax

${VM_DATE} : The date and time when the Fax is received. (Format: MM/dd/yyyy hh:mm:ss

Send PDF Files

If enabled, fax emails will no longer attach TIFF files. Only PDF files will be attached.
Only

Enables the fax resend option which allows the UCM to keep attempting to send faxes up to a specified amount of times. Additionally,
Enable Fax
if fax still fails to send, a Resend button will appear in the File Send Progress list in Other Features🡪Fax Sending to allow manual
Resend
resending.

Configures the number of the maximum attempts to resend the fax.

Max Resend

Attempts
The default value is set to 5.

Configures the Fax Resend Frequency.

Fax Resend

Frequency
The default value is set to 50.

Table 71: FAX/T.38 Settings

Receiving Fax

Example Configuration to Receive Fax from PSTN Line

The following instructions describe how to use the UCM6300 to receive fax from the PSTN line on the Fax machine connected to the UCM6300 FXS port.

1. Connect Fax machine to the UCM6300 FXS port.

 2. Connect PSTN line to the UCM6300 FXO port.

3. Go to Web GUI🡪Extension/Trunk page.

4. Create or edit the analog trunk for Fax as below.

Fax Detection: Make sure the “Fax Detection” option is set to “NO”.

Configure Analog Trunk

5. Go to UCM6300 Web GUI🡪Extension/Trunk🡪Extensions page.

6. Create or edit the extension for FXS port.

Analog Station: Select FXS port to be assigned to the extension. By default, it is set to “None”.

Once selected, analog related settings for this extension will show up in “Analog Settings” section.

Configure Extension for Fax Machine: FXS Extension

 Configure Extension for Fax Machine: Analog Settings

7. Go to Web GUI🡪Extension/Trunk🡪Inbound Routes page.

8. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in step 4 as the default destination.

Configure Inbound Rule for Fax

Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO port, it will send the Fax to the Fax machine.

Example Configuration for Fax-To-Email

The following instructions describe a sample configuration on how to use Fax-to-Email feature on the UCM6300.

1. Connect PSTN line to the UCM6300 FXO port.

2. Go to UCM6300 Web GUI🡪Call Features🡪Fax/T.38 page. Create a new Fax extension.

 Create Fax Extension

3. Go to UCM6300 Web GUI🡪Extension/Trunk🡪Analog Trunks page. Create a new analog trunk. Please make sure “Fax Detection” is set to “No”.

4. Go to UCM6300 Web GUI🡪Extension/Trunk🡪Inbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

Inbound Route to Fax Extension

5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF+Tiff file and sent to the

extension 7200 and email address fax@domain.local as attachment.

In order for the file to be sent to the email address configured on the external extension, please make sure that the email settings are well configured. Please

refer to [Email Settings] section.

List of Fax Files

FAX Sending
Besides the support of Fax machines, The UCM6300 supports also sending Fax via Web GUI access. This feature can be found on Web GUI🡪Other

Features🡪Fax Sending page. To send fax, pre-setup for analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP
TRUNKS] and [Outbound Routes] sections for configuring analog trunk and outbound route.

After making sure analog trunk or VoIP Trunk is setup properly and UCM6300 can reach out to PSTN numbers via the trunk, on Fax Sending page, enter the

fax number and upload the file to be faxed. Then click on “Send” to start. The progress of sending fax will be displayed in Web GUI. Users can also view the
sending history is in the same web page.

Fax Sending in Web GUI

After that you can see the ongoing sending operation on the progress bar.

Fax Send Progress

Only A3, A4, and B4 paper sizes are supported for the Fax Sending.

MULTIMEDIA MEETING
The UCM630X supports multimedia meeting room allowing multiple rooms used at the same time.

The multimedia meeting room configurations can be accessed under Web GUI🡪Call Features🡪 Multimedia Meeting. In this page, users could create, edit,

view, invite, manage the participants, and delete multimedia meeting rooms. The multimedia meeting room status and meeting call recordings (if recording is
enabled) will be displayed in this web page as well.

For video meeting, which is based on WebRTC, participants can join the meeting from PC without installing extra plug-in or software.

The UCM admin can create multiple multimedia meeting rooms for users to dial in.
 UCM630x Series Number of public meeting rooms Number of meeting room members

UCM6301 4 75

UCM6302 8 150

UCM6304 15 200

UCM6308 25 300

Meeting room specifications affect user participation to a certain extent. UCM supports forecasting of meeting resources. There will be corresponding

judgments and adjustments in the following scenarios:

1. When meeting resources are used up, scheduled meeting members cannot join the meeting in advance.

2. When a point-to-point call is transferred to a conference, the conference resources are used up.

3. When meeting resources are used up, do not join a group IM chat when you initiate a meeting.

4. When meeting resources are used up, do not join an instant meeting.

5. Close other instant meetings or scheduled meetings that have timed out to ensure that invited members can join the scheduled meeting.

6. In an ongoing meeting, if the number of invited members exceeds the upper limit, members cannot be invited to join the meeting.

7. Enable flow control for videos and presentations in the conference room.

Notes:

The multimedia meeting room supports up to 4 video calls and one video presentation.

– The administrator can set the number of videos to 9 parties. The increase in the number of videos will take up more system resources and affect the overall

performance of the UCM system. Please set it according to your needs.

– During a meeting, when the system detects that another scheduled meeting is about to be held, it will remind the meeting members that the subsequent

meeting room has been reserved, please end the meeting in advance.

– The use of video in the meeting will take up system resources and may cause performance problems when used.

– The maximum meeting duration is 12 hours. If it exceeds 12 hours, the system will remind the current meeting and the host can continue to extend the

meeting.

Multimedia Room Configurations

Click on “Add” to add a new meeting room.

Click on to edit the meeting room.

Click on to delete the meeting room.

Meeting Settings contains the following options:

Extension The number to dial to enter the meeting room.

 Meeting Name Enter the meeting's name.

Privilege Select the permission level for outgoing call.

Allow User Invite If enabled, participants can invite other members to the meeting room.

Allowed to Override
When turned on, participants are allowed to actively unmute when they are muted by the moderator.
Host Mute

Select an option to record the meeting. The recording options are:

● None: Meeting recording is disabled.

● Record Audio: Only the audio of the meeting will be recorded.

● Record Video: Both audio and video recordings of the meeting will be recorded.

● Record Video (Only Shared Screen): Only shared screen video feed will be recorded and the audio of all the participants in the
Auto Record meeting. Please note that the recording will only start when the screensharing starts. Any video feeds already ongoing will not be

recorded.

The recording audio files are stored as .WAV files, while the audio-video recording files are stored as .MKV.

The files names are created following this format: meeting-${Meeting Number}-${UNIQUEID}. The recordings can be downloaded

from the Meeting Recordings menu or the Meeting Video Recordings page.

Note: When this option has been enabled the meeting host cannot stop the recording of the meeting.

If meeting room password is configured, meeting participants will need to enter a password to enter the room. Scheduling meetings
Room Password
will not be supported for this room.

Log in to the UCM Web GUI and open the Call Features 🡪 Multimedia Meeting page to manage the conference room. Users can create, edit, view, invite,

manage meeting members, and delete meeting rooms. The conference room status and conference call recording (if recording function is enabled) will be
displayed on the page. The meeting rooms in the list include public meeting rooms and random meeting rooms. For temporary meeting room administrators,
only the “batch kicking people” function is supported. The temporary meeting room has no meeting password and host code. The member who initiates the
group meeting is the host, and ordinary members have the right to invite.

Multimedia Meeting
Multimedia Meeting Call Operations

Join a Meeting Call

Users could dial the meeting room extension to join the meeting. If password is required, enter the password to join the meeting as a normal user, or enter the

admin password to join the meeting as administrator.

Invite Other Parties to Join Meeting

When using the UCM630X meeting room., there are two ways to invite other parties to join the meeting.

Invite from Web GUI.

For each meeting room in UCM630X Web GUI🡪Call Features🡪 Multimedia Meeting, there is an icon for option “Invite a participant”. Click on it and
enter the number of the party you would like to invite. Then click on “Add”. A call will be sent to this number to join it into the conference.

Meeting Invitation from Web GUI

Invite by dialing 0 or 1 during conference call.

A meeting participant can invite other parties to the meeting by dialing from the phone during the meeting call. Please make sure option “Enable User Invite”

is turned on for the meeting room first. Enter 0 or 1 during the meeting call. Follow the voice prompt to input the number of the party you would like to invite.
A call will be sent to this number to join it into the meeting.

0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be asked to “accept” or “reject” the invitation before

joining the conference.

1: If 1 is entered to invite other party, no permission will be required from the invited party.

Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To join a conference room as administrator, enter the

admin password when joining the conference. A conference room can have multiple administrators.

During the meeting

During the meeting call, users can manage the conference from Web GUI or IVR.

Manage the meeting call from Web GUI.

Log in UCM630X Web GUI during the meeting call, the participants in each meeting room will be listed.

1. Click on to kick a participant from the meeting.

2. Click on to mute the participant.

 3. Click on to lock this meeting room so that other users cannot join it anymore.

4. Click on to invite other users into the meeting room.

5. Click on to Invite meeting rooms or invite contact groups.

Manage the meeting call from IVR.

Please see options listed in the table below.

Meeting Administrator IVR Menu

1 Mute/unmute yourself.

2 Lock/unlock the conference room.

3 Kick the last joined user from the conference.

4 Decrease the volume of the conference call.

5 Decrease your volume.

6 Increase the volume of the conference call.

7 Increase your volume.

More options.

1: List all users currently in the conference call.

2: Kick all non-Administrator participants from the conference call.

8
3: Mute/Unmute all non-Administrator participants from the conference call.

4: Record the conference call.

8: Exit the caller menu and return to the conference.

Meeting User IVR Menu

1 Mute/unmute yourself.

4 Decrease the volume of the conference call.

5 Decrease your volume.

6 Increase the volume of the conference call.

7 Increase your volume.

8 Exit the caller menu and return to the conference.

 Table 74: Meeting Caller IVR Menu

When there is participant in the meeting, the meeting room configuration cannot be modified.

Google Service Settings Support

UCM630X now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM630X conference scheduling system. Once OAuth 2.0 is

enabled, UCM630X conference system can access Google calendar to schedule or update conference.

Google Service Settings can be found under Web GUI🡪Call Features🡪 Multimedia Meeting 🡪Google Service Settings🡪Google Service Settings.

Google Service Settings🡪OAuth2.0 Authentication

If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login credential for “OAuth2.0 Client ID” and

“OAuth2.0 Client Secret” in the above figure for the UCM630X to access Google Service.

If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain credentials:

1. Go to Google Developers page https://console.developers.google.com/start Create a New Project in Google Developers page.

Google Service🡪New Project

2. Enable Calendar API from API Library.

3. Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials.
 Google Service🡪Create New Credential

4. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.

5. Click “Get Authentication Code” to obtain authentication code from Google Service.

Google Service🡪OAuth2.0 Login

6. Now UCM630X is connected with Google Service.

You can also configure the Status update, which automatically refresh your Google Calendar with the configured time (m). Note: Zero means disable.

Meeting Schedule
Log in to the UCM Web GUI, open the Call Features 🡪 Mutimedia Meeting 🡪 Meeting Schedule page, and you can manage the reservation management

of the meeting room. Users can create, edit, view, and delete conference room reservation records. The following is a set meeting room reservation, which

shows the ongoing and pending reservations. Once the conference room is reserved, all users will be removed from the conference room at the start time, and

extensions will no longer be allowed to enter the conference room. At the scheduled meeting time, UCM will send invitations to the extensions that have been

selected to participate in the meeting. At the same time, it supports users to enter the meeting 10 minutes in advance. If the current meeting is occupied, enter

the waiting room and wait (members joining the meeting in advance occupy global member resources, but it will be released after the scheduled meeting

starts); otherwise, you can join the meeting directly and the meeting will be held in advance. After the meeting ends, the reservation record is transferred to

the historical meeting list. History meeting displays the information of the ended and expired meetings.
 Click the button “Schedule Meeting” to edit the meeting room reservation.

Schedule meeting Interface

Schedule Options

Configure the name of the scheduled conference. Letters, digits, Other special characters are also supported. such as
Meeting Subject
#%&@*=

Choose which room to have this scheduled meeting.

Meeting Room If this option has been enabled, please select an existing room for this meeting. If this option has not been enabled, a new

meeting room will be created.

Time Configure the meeting date and time.

Time Zone Select the meeting time zone.

Password Configure the conference login password.

Configure the Host Password.

Host Password
Note: It is randomly generated when first creating a new meeting Schedule.

Host Configure Host.

Repeat Choose when to repeat a scheduled conference.

If this option is enabled, the user can:

● Press ‘0’ to invite others to join the meeting with invited party’s permission

● Press ‘1’ to invite without invited party’s permission

Allow User Invite ● Press ‘2’ to create a multi-meeting room to another meeting room

● Press ‘3’ to drop all current multi-meeting rooms.

Note:

Meeting host is always allowed to access this menu.

 Call Participant If enabled, the invited participants will be called upon meeting start time.

Allowed to Override Host Mute If enabled, participants will be able to unmute themselves if they have been muted by the host.

Email reminders will be sent out x minutes prior to the start of the meeting. Valid range is 5-1440. 60 is the default value.

0 indicates not to send out email reminders for the meeting.

Email Reminder (m)
Note: After editing the time of a single recurrence of a scheduled meeting, a cancelation email will now be sent out

followed by a meeting update email.

If selected, the meeting will be recorded and saved as either a .WAV or .MKV file. The default filename

is meeting-${Meeting Number}-${UNIQUEID}. Recordings can be downloaded from either the Meeting Recordings or

Auto Record the Meeting Video Recordings page. Video recordings require external storage to be available. When recording a screen

share, only the screen share and meeting audio will be recorded.

Note: Please note that UCM63XX Audio Series doesn't support Screen Sharing, Whiteboard, or PDF file sharing.

Select this option to sync scheduled conference with Google Calendar.

Enable Google Calendar Note: Google Service Setting OAuth2.0 must be configured on the UCM630X. Please refer to section [Google Service

Settings Support].

Meeting Agenda Enter information about the meeting, e.g., the purpose of the meeting or the subjects that will be discussed in the meeting.

Invitees Local extensions, remote extensions, and special extensions are supported.

Once the Meeting Schedule is configured, scheduled meeting will be displayed as below figure.

Meetings Schedule

Click the button to view the meeting details in the Meeting room. The meeting details of Meeting History include actual participant information.
 Meeting details

Click on to edit the Meeting Schedule.

Click on to delete the Meeting Schedule.

At the scheduled meeting time, UCM630X will send INVITE to the extensions that have been selected for conference.

Once the meeting starts, it will be displayed under Pending Meeting with an “Ongoing” status, as displayed below:

Figure 152: Meeting Scheduled – Ongoing

Once the conference is finished, the conference will be displayed under Historical meeting as below:

Meeting Schedule – Completed

 Click the button to download the Meeting Report of the meeting.

Click the button to reschedule the Meeting.

In addition, once the meeting ends, the system will send a meeting report email to the host including PDF file where he/she can view the meeting, participant

information, device type and trend graph of participant levels.

You can also choose to display the meetings that took place in a specific timeframe. Please see the screenshot below:

Please make sure that outbound route is properly configured for remote extensions to join the meeting.

Meeting Recordings
The UCM630X allows users to record the audio of the meeting call and retrieve the recording from Web GUI🡪Call Features🡪 Multimedia Meeting🡪
Meeting Recordings.

To record the Meeting call, when the meeting room is in idle, enable “Auto Record” from the meeting room configuration dialog. Save the setting and apply

the change. When the meeting call starts, the call will be automatically recorded in .wav format.

The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. Users

could also delete all recording files by clicking on “Delete All Recording Files” or delete multiple recording files at once by clicking on “Delete” after

selecting the recording files.

Meeting Recordings
Meeting Video Recordings
The UCM630X allows users to record the audio and video of the meeting call and retrieve the recording from Web GUI🡪Call Features🡪 Multimedia

Meeting🡪 Meeting Recordings.

To record the Meeting call, when the meeting room is in idle, enable “Auto Record” from the meeting room configuration dialog. Save the setting and apply

the change. When the meeting call starts, the call will be automatically recorded in .mkv format.

The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. Users

could also delete all recording files by clicking on “Delete All Recording Files” or delete multiple recording files at once by clicking on “Delete” after

selecting the recording files.

Call Statistics
Meeting reports will now be generated after every conference. These reports can be exported to a .CSV file for offline viewing. The conference report page

can be accessed by clicking on the Call Statistics button on the main Conference page.

Meeting Call Statistics

Meeting Report on Web

Meeting Report on CSV

ONSITE MEETING
For workplaces that require employees to return to physical offices for work, Grandstream UCM offers the Onsite Meetings feature, a new way to stay
organized and keep up-to-date with in-person meetings. This feature allows administrators to create and manage onsite meeting rooms, specify meeting room

locations, schedule meetings, and add conferencing equipment. The new feature can be found under the Other Features🡪Onsite Meeting page. The first page

that appears is the Scheduled Meetings page and tab page, which provide an overview of all created meeting rooms. It provides information about the rooms’
meeting schedules for the day, their locations, their member capacity, and their equipment.
 Schedule Onsite Meetings

The Pending Meeting tab and Meeting History tab show detailed information about upcoming meetings and previous meetings respectively. From the

Pending Meeting tab, users can delete upcoming meetings and extend the duration of ongoing meetings. The Meeting History tab will display the last 6

months of onsite meetings.

Pending Onsite Meetings

IVR

Configure IVR
IVR configurations can be accessed under the UCM630X Web GUI🡪Call Features🡪IVR. Users could create, edit, view, and delete an IVR.

Click on “Add” to add a new IVR.

Click on to edit the IVR configuration.

Click on to delete the IVR.

 Create New IVR

Basic

Settings

Name Configure the name of the IVR. Letters, digits, _ and – are allowed.

Extension Enter the extension number for users to access the IVR.

If enabled, all callers to the IVR can use trunk. The permission must be configured for the users to use the trunk first. The default setting is
Dial Trunk
“No”.

Auto
If enabled, calls to this IVR will automatically be recorded.
Record

Assign permission level for outbound calls if “Dial Trunk” is enabled. The available permissions are “Internal”, “Local”, “National” and

“International” from the lowest level to the highest level.

Permission The default setting is “Internal”. If the user tries to dial outbound calls after dialing into the IVR, the UCM630X will compared the IVR’s

permission level with the outbound route’s privilege level.

If the IVR’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through.
 This controls the destination that can be reached by the external caller via the inbound route. The available destinations are:

Extension

Conference

Call Queue
Dial Other
Ring Group
Extensions

Paging/Intercom Groups

Voicemail Groups

Dial by Name

All

IVR

Black/Whit If enabled only numbers inside of the Whitelist or outside of the Blacklist can be called from IVR.

elist

Internal
Black/Whit Contain numbers, either of Blacklist or Whitelist.
elist

External
This feature can be used only when Dial Trunk is enabled, it contains external numbers allowed or denied calling from the IVR, the allowed
Black/Whit
format is the following: Number1, number2, number3…
elist

Replace

Display If enabled, the UCM will replace the caller display name with IVR name.

Name

Return to
If enabled and if a call to an extension fails, the caller will be redirected to the IVR menu.
IVR Menu

Alert Info When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS.

Select an audio file to play as the welcome prompt for the IVR. Click on “Prompt” to add additional audio file under Web GUI🡪PBX
Prompt
Settings🡪Voice Prompt🡪Custom Prompt.

Digit Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit
Timeout is detected within the timeout, the UCM630X will consider the entries complete. The default timeout is 3s.

Response After playing the prompts in the IVR, the UCM630X will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is

Timeout detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds.

Response

Timeout Select the prompt message to be played when timeout occurs.

Prompt
Invalid
Input Select the prompt message to be played when an invalid extension is pressed.

Prompt

Response
Timeout Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination
Prompt if configured, or hang up. The default setting is 3.

Repeats

Invalid

Input Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if
Prompt configured, or hang up. The default setting is 3.
Repeats

Select the voice prompt language to be used for this IVR. The default setting is “Default” which is the selected voice prompt language under

Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the current available voice prompt languages
Language
on the UCM630X. To add more languages in the list, please download voice prompt package by selecting “Check Prompt List” under Web

GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Key

Pressing

Events
 Select the event for each key pressing for 0-9, *, Timeout and Invalid. The event options are:

Extension

Key Press Voicemail

Event:
Multimedia Meeting

Press 0 Voicemail Group

Press 1 IVR

Ring Group
Press 2
Queues
Press 3
Page Group

Press 4 Custom Prompt

Hangup
Press 5
DISA
Press 6
Dial by Name

Press 7 External Number

Press 8 Callback

Press 9 For each key event, time condition can be configured. At the configured time condition, this IVR key event can be triggered. Office time,

holiday time or specific time can be configured for time condition. Up to 5 time conditions can be added for each key.
Press *
The available time conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’,
‘Office Time and Out of Holiday’ and ‘Specific Time’. If ‘Specific Time’ is selected, a new window will prompt for admin to configure start

time, end time and frequency.

When exceeding the number of defined answer timeout, IVR will enter the configured event when timeout. If not configured, then it will
Timeout
Hangup.

Invalid Configure the destination when the Invalid Repeat Loop is done.

Time
Configure the time condition for each key press event, so that it goes to the corresponding destination within a specified time.
Condition

Table 76: IVR Configuration Parameters

 Figure 159: Key Pressing Events

Black/Whitelist in IVR
In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.

For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some special extensions are used on IP surveillance end
points which should not be reached from external calls via IVR for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to

manage this.

Up to 500 extensions are allowed on the back/whitelist.

To use this feature, log in UCM Web GUI and navigate to Call Features🡪IVR🡪Create/Edit IVR: IVR Black/Whitelist.

If the user selects “Blacklist Enable” and adds extension in the list, the extensions in the list will not be allowed to be reached via IVR.

If the user selects “Whitelist Enable” and adds extension in the list, only the extensions in the list can be allowed to be reached via IVR.
 Black/Whitelist

Create Custom Prompt

To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Upload Audio File” next to the “Welcome Prompt” option and the users will be

redirected to Custom Prompt page. Or users could go to Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page directly.

Click on Prompt to Create IVR Prompt

Once the IVR prompt file is successfully added to the UCM630X, it will be added into the prompt list options for users to select in different IVR scenarios.

Key Pressing Events

Standard Key Event

UCM supports adding time conditions for different key events, so that each key event of the IVR goes to the corresponding destination within a specified

time.

Each key event support up to five time conditions, the options available are: All time, Office Time, Out of Office Time, Holiday, Out of Holiday, Out of Office

Time or Holiday, Office Time and Out Of Holiday, Specific time.

 Key Pressing Events

Note:

If you select “Specific time”, you need to select the start time and the end time.

The frequency supports two options: By week and By Month, by default the specific time does not include the holidays.

Specific Time

Custom Key Event

Users can create custom IVR key press events, vastly increasing the options a business can provide to its customers and improving customer relations and

accessibility.
This new feature supports the following:

Up to 100 custom key press events

Each key combination can contain up to 8 characters (numbers and star (*) only)

Supports Time Conditions

Different custom keys can have the same Destination and Time Condition

Note

Note: IVR option Dial Other Extensions will be disabled if using custom IVR keys.

LANGUAGE SETTINGS FOR VOICE PROMPT

The UCM630X supports multiple languages in Web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt:

English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian,

Spanish, Catalan, Swedish and Turkish.

English (United States) and Chinese voice prompts are built in with the UCM630X already. The other languages provided by Grandstream can be downloaded

and installed from the UCM630X Web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the
UCM630X.

Language settings for voice prompt can be accessed under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Download and Install Voice Prompt Package

To download and install voice prompt package in different languages from UCM630X Web GUI, click on “Add Voice Prompt Package” button.
 Language Settings for Voice Prompt

A new dialog window of voice prompt package list will be displayed. Users can see the version number (latest version available V.S. current installed

version), package size and options to upgrade or download the language.

Voice Prompt Package List

Click on to download the language to the UCM630X. The installation will be automatically started once the downloading is finished.
 New Voice Prompt Language Added

A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM630X system voice prompt or delete it
from the UCM630X.

Customize Specific Prompt

On the UCM630X, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from Web

GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings and click on “Upload” instead of the entire language pack.

Upload Single Voice Prompt for Entire Language Pack

Username Prompt Customization

There are two ways to customize/set new username prompt:

Upload Username Prompt File from Web GUI

1. First, Users should have a pre-recorded file respecting the following format:

PCM encoded / 16 bits / 8000Hz mono.

In .tar/.tar.gz/.tgz format

File size under 30M.

Filename must be set as the extension number with 18 characters max. For example, the recorded file name 1000.wav will be used for extension 1000.
 2. Go under web GUI PBX Settings 🡪 Voice Prompt 🡪 Username Prompt and click on ”Upload” button.

3. Select the recorded file to upload it and press Save and Apply Settings.

Click on to record again the username prompt.

Click on to play recorded username prompt.

Select username prompts and press to delete specific file or select multiple files for deletion using the button ”Delete” .

Record Username via Voicemail Menu

The second option to record username is using voicemail menu, please follow below steps:

Dial *98 to access the voicemail

After entering the desired extension and voicemail password, dial “0” to enter the recordings menu and then “3” to record a name.

Another option is that each user can record their own name by following below steps:

The user dials *97 to access his/her voicemail

After entering the voicemail password, the user can press “0” to enter the recordings menu and then “3” to record his name.

VOICEMAIL

Configure Voicemail
If the voicemail is enabled for UCM630X extensions, the configurations of the voicemail can be globally set up and managed under Web GUI🡪Call

Features🡪Voicemail.

Voicemail Settings
Max Greeting Time
Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds.
(s)

Dial ‘0’ For Operator If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator’s extension.

Operator Type Configure the operator type; either an extension or a ring group.

Select the operator extension, which will be dialed when users press 0 to exit voicemail application. The operator extension can
Operator Extension
also be used in IVR.

Max Messages Per

Configure the maximum number of messages per folder in users’ voicemail. The valid range 10 to 1000. The default setting is 50.
Folder

Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max

message time. The default setting is 15 minutes. The available options are:

1 minute

2 minutes
Max Message Time
5 minutes

15 minutes

30 minutes

Unlimited

Configure the minimum duration (in seconds) of a voicemail message. Messages will be automatically deleted if the duration is

shorter than the Min Message Time. The default setting is 3 seconds. The available options are:

No minimum

1 second
Min Effective
2 seconds
Message Time
3 seconds

4 seconds

5 seconds

Note: Silence and noise duration are not counted in message time.

Announce Message If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The
Caller-ID default setting is “No”.

Announce Message
If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is “No”.
Duration

If enabled, a brief introduction (received time, received from, and etc.) of each message will be played when accessed from the
Play Envelope
voicemail application. The default setting is “Yes”.
 Play Most Recent
If enabled, it will play the most recent message first.
First

Allow User Review If enabled, users can review the message following the IVR before sending.

If enabled, external callers routed by DID and reaching VM will be prompted by the UCM with 2 options:

Press 1 to leave a message.

To leave a message for the extension reached by DID.

Voicemail Remote
Press 2 to access voicemail management system.
Access

This will allow caller to access any extension VM after entering extension number and its VM password.

Note: This option applies to inbound call routed by DID only.

The default setting is “Disabled”.

Enables the forwarding of voicemail to remote extensions on peered SIP trunks.

Forward Voicemail

to Peered UCMs
The default setting is “Disabled”.

Voicemail Password Configures the default voicemail password that will be used when an extension is reset.

Format Warning: WAV files take up significantly more storage space than GSM files.

Table 77: Voicemail Settings

Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to default. Previous custom voicemail
prompts and messages will be deleted.

Access Voicemail
If the voicemail is enabled for UCM630X extensions, the users can dial the voicemail access number (by default *97) to access their extension’s voicemail.

The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different
options.

Otherwise the user can dial the voicemail access code (by default *98) followed by the extension number and password in order to access to that specific
extension’s voicemail.

Main Menu Sub Menu 1 Sub Menu 2

1 – New messages 3 - Advanced options 1 - Send a reply

2 - Call the person who sent this message

 3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

5 - Repeat the current message

7 - Delete this message

8 - Forward the message to another user

9 – Save

* - Help

# - Exit

0 - New messages

1 - Old messages

2 - Work messages

2 – Change folders

3 - Family messages

4 - Friend messages

# - Cancel

1 - Send a reply

2 - Call the person who sent this message

3 – Advanced options 3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

0 – Mailbox options 1 - Accept this recording

1 - Record your unavailable message 2 - Listen to it

3 - Re-record your message

2 - Record your busy message 1 - Accept this recording

 2 - Listen to it

3 - Re-record your message

1 - Accept this recording

3 - Record your name 2 - Listen to it

3 - Re-record your message

1 - Accept this recording

4 - Record temporary greeting 2 - Listen to it

3 - Re-record your message

5 - Change your password

* - Return to the main menu

Tips

While listening to the voicemail, press * or # to rewind and forward the voice message, respectively. Each press will forward or rewind 3 seconds.

Rewind can go back to the begining of the message while forward will not work when there are 3 seconds or less left in the voice message.

Voice guidance will be automatically played when the voicemail is done playing.

Leaving Voicemail
If an extension has voicemail enabled under basic settings “Extension/Trunk 🡪 Extensions 🡪 Basic Settings” and after a ring timeout or user not available,

the caller will be automatically redirected to the voicemail in order to leave a message on which case they can press # in order to submit the message.

In case if the caller is calling from an internal extension, they will be directly forwarded to the extension’s voicemail box. But if the caller is calling from

outside the system and the incoming call is routed by DID to the destination extension, then the caller will be prompted with the choice to either press1 to

access voicemail management or press 2 to leave a message for the called extension. This feature could be useful for remote voicemail administration.

Voicemail Email Settings

The UCM630X can be configured to send the voicemail as attachment to Email. Click on “Voicemail Email Settings” button to configure the Email attributes

and content.

Send Voicemail to Email If enabled, voicemail will be sent to the user’s email address. Note: SMTP server must be configured to use this option.

Keep Voicemail after

Enable this option if you want to keep recording files after the Email is sent. The default setting is Enable.
Emailing
 Fill in the “Subject:” and “Message:” content, to be used in the Email when sending to the user. The template variables
are:

\t: TAB

${VM_NAME}: Recipient’s first name and last name

Email Template ${VM_DUR}: The duration of the voicemail message

${VM_MAILBOX}: The recipient’s extension

${VM_CALLERID}: The caller ID of the person who has left the message

${VM_MSGNUM}: The number of messages in the mailbox

${VM_DATE}: The date and time when the message is left. (Format: MM/dd/yyyy hh:mm:ss)

Table 79: Voicemail Email Settings

Voicemail Email Settings

Click on “Email Template” button to view the default template as an example.

Configure Voicemail Group

The UCM630X supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group

can be configured under Web GUI 🡪 Call Features 🡪 Voicemail 🡪 Voicemail Group. Click on “Add” to configure the group.
 Voicemail Group

Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group
Extension
members.

Name Configure the Name to identify the voicemail group. Letters, digits, _ and - are allowed.

Voicemail Password Configure the voicemail password for the users to check voicemail messages.

Email Address Configure the Email address for the voicemail group extension.

Shared Voicemail If enabled, voicemail group status can be monitored via BLF. Green indicates no unread voicemail, and red indicates existing

Status unread voicemail.

Select available mailboxes from the left list and add them to the right list. The extensions need to have voicemail enabled to be
Member
listed in available mailboxes list.

Busy Prompt This voicemail prompt will be played when the callee is in another call or when he/she is in DND mode. Priority: Temporary

Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

 Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt will be played when the callee does not answer within their ring timeout period. Priority: Temporary Prompt

> Busy Prompt/Unavailable Prompt > Greet Prompt

Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt will be played in all scenarios when it is configured (unregistered, unanswered/ring timeout, busy, DND).

Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Temporary Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt will be played when user enters voicemail. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt

> Greet Prompt

Unavailable Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

RING GROUP
The UCM630X supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group

configuration on the UCM630X.

Configure Ring Group

Ring group settings can be accessed via Web GUI🡪Call Features🡪Ring Group.

Ring Group

Click on to add ring group.

Click on to edit the ring group. The following table shows the ring group configuration parameters.

Click on to delete the ring group.

Ring Group Name Configure ring group name to identify the ring group. Letters, digits, _ and – are allowed.

Extension Configure the ring group extension.

Members Select available users from the left side to the ring group member list on the right side. Click on ⮝ ⮟ to arrange the order.
 Select available remote users from the left side to the ring group member list on the right side. Click on ⮝ ⮟ to arrange the order.
LDAP Phonebook
Note: LDAP Sync must be enabled first.

Select the ring strategy. The default setting is “Ring in order”.

● Ring Simultaneously: Ring all the members at the same time when there is incoming call to the ring group extension. If any of
Ring Strategy the member answers the call, it will stop ringing.

● Ring in Order: Ring the members with the order configured in ring group list. If the first member does not answer the call, it
will stop ringing the first member and start ringing the second member.

Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be managed from the “Music On Hold” panel on the
Music On Hold
left.

This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the

customized voice prompts.

Custom Prompt
Note: Users can also refer to the page PBX Settings🡲 Voice Prompt🡲 Custom Prompt, where they could record new prompt or

upload prompt files.

Ring Timeout on Each Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 60 seconds.

Member Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.

If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed
Auto Record
from WebGUI🡲 CDR🡲 Recording Files.

This allows the UCM to work with endpoint-configured call forwarding settings to redirect calls to ring group. For example, if a

member wants to receive calls to the ring group on his mobile phone, he will have to set his endpoint’s call forwarding settings to

his mobile number. By default, it is disabled.

However, this feature has the following limitations:

Endpoint Call ● This feature will work only when call forwarding is configured on endpoints, not on the UCM.
Forwarding Support ● If the forwarded call goes through an analog trunk, and polarity reversal is disabled, the other ring group members will no longer
receive the call after it is forwarded.

● If the forwarded call goes through a VoIP trunk, and the outbound route for it is PIN-protected and requires authentication, the
other ring group members will no longer receive the call after it is forwarded.

● If the forwarded call hits voicemail, the other ring group members will no longer receive the call.

If enabled, the UCM will replace the caller display name with the Ring Group name the caller know whether the call is incoming
Replace Display Name
from a direct extension or a Ring Group.

Skip Busy Agent If enabled, skip busy agents regardless of call waiting settings.

If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the
Enable Destination
ring group has no answer. Secret and Email address are required if voicemail is selected as the destination.

The call would be routed to this destination if no one in this ring group answers the call.

Default Destination Note: Users can now set the voicemail of ring groups as routing destinations and IVR key press event destinations and to do so ring

group must have their Default Destination set to Voicemail with Ring Group Extensions.
Voicemail Whether to enable the voicemail for the ring group or not.

Voicemail Password Configure the voicemail password (only numbers).

Email Address Fill in the user's Email address (s), the voice message will be sent to this address (s).

This voicemail prompt will be played when the callee is in another call or is in DND mode. Priority: Temporary Prompt > Busy

Prompt/Unavailable Prompt > Greet Prompt

Busy Prompt
Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt will be played when the callee does not answer within their ring timeout period. Priority: Temporary Prompt

> Busy Prompt/Unavailable Prompt > Greet Prompt

Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt well be played in all scenarios when it is configured (unregistered, unanswered/ring timeout, busy, DND).

Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Temporary Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

This voicemail prompt will only be played when the callee’s extension is unregistered. Priority: Temporary Prompt > Busy

Prompt/Unavailable Prompt > Greet Prompt

Unavailable Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with

.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

 Ring Group Configuration

Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM630X can be included in the ring group with local extension. An example of Ring Group with peer

extensions is presented in the following:

1. Creating SIP Peer Trunk between both UCM630X _A and UCM630X _B. SIP Trunk can be found under Web GUI🡪Extension/Trunk🡪VoIP Trunks.

Also, please configure their Inbound/Outbound routes accordingly.

2. Click edit button in the menu , and check if Sync LDAP Enable is selected, this option will allow UCM630X_A update remote LDAP server

automatically from peer UCM630X _B. In addition, Sync LDAP Password must match for UCM630X _A and UCM630X _B to sync LDAP contact

automatically. Port number can be anything between 0~65535, and use the outbound rule created in step 1 for the LDAP Outbound Rule option.
 Sync LDAP Server Options

3. In case if LDAP server does not sync automatically, user can manually sync LDAP server. Under VoIP Trunks page, click sync button shown in the

following figure to manually sync LDAP contacts from peer UCM630X.

Manually Sync LDAP Server

4. Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI🡪Call Features🡪Ring Groups.

5. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring

group. Please also make sure the extensions in the peer UCM630X can be included into that UCM630X’s LDAP contact.

Ring Group Remote Extension

RESTRICT CALLS
Restrict calls is a feature that can be used to restrict calls between internal extensions besides those in the Allowed List.

This section describes the configuration of this feature in the Call Features->Restrict Calls page.
 Restrict Calls

Configure Restrict Calls

Click on “Add” to add a rule for restrict calls.

Click on to edit the rule of restrict calls.

Click on to delete the rule of restrict calls.

Name Configure Restrict call’s name

Restrict Calls between When enabled, members of the group cannot dial other extension, only the numbers in the Allowed List. By default it’s

extensions enabled.

Members Configure the members that will not be able to call any extensions besides those in the Allowed List.

Allowed list Select the extensions that the Members list can be able to call.

PAGING AND INTERCOM GROUP

Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones. Targeted phones will answer immediately using
speaker. The UCM630X paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the

configuration of paging/intercom group under Web GUI🡪Call Features🡪Paging/Intercom.

Configure Paging/Intercom Group

Click on “Add” to add paging/intercom group.

Click on to edit the paging/intercom group.

Click on to delete the paging/intercom group.

 Click on “Paging/Intercom Group Settings” to edit Alert-Info Header. This header will be included in the SIP INVITE message sent to the callee in

paging/intercom call.

Configure Multicast Paging

Multicast Paging

Name Configure paging/intercom group name.

Type Select “Multicast Paging”.

Extension Configure the paging/intercom group extension.

If enabled, a caller can enter *82 before the paging group extension to start a delayed paging call. In a delayed paging call, the

system will prompt the caller to record a message. Once the messaging is recorded and saved, and the configured delay has
Delayed Paging
passed, the paging call will be sent out. When a paging group member answers the call, the recorded message will be played, and

the call will end after it is finished playing.

Delay (s) Configure the amount of delay in seconds after a message is recorded to send out the delayed paging call. Default is 5 seconds.

The allowed multicast IP address range is 224.0.1.0 – 238.255.255.255.

Multicast IP Address
Note: This field appears only when “Type” is set to “Multicast Paging”.

Maximum Call Duration Specify the maximum call duration in seconds. The default value 0 means no limit.

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users

to upload the customized voice prompts.

Custom Prompt
Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or

upload prompt files.

Multicast IP Address Select which extensions are allowed to use the paging/intercom feature for this paging group.

Specify port for multicast paging.

Port
Note: This field appears only when “Type” is set to “Multicast Paging”.
Configure 2-way Intercom

2-way Intercom

Name Configure paging/intercom group name.

Type Select “2-way Intercom”.

Extension Configure the paging/intercom group extension.

Auto Record Enable this option to record in WAV format.

Replace Display
If enabled, the UCM will replace the caller display name with Paging/Intercom name.
Name

Maximum Call
Specify the maximum call duration in seconds. The default value 0 means no limit.
Duration

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users

to upload the customized voice prompts.

Custom Prompt
Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or
upload prompt files.

Members Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom
Select which extensions are allowed to use the paging/intercom feature for this paging group.
Whitelist
 Table 83: 2-way Intercom Configuration Parameters

Configure 1-way Paging

1-way Paging

Name Configure paging/intercom group name.

Type Select “1-way Paging”.

Extension Configure the paging/intercom group extension.

If checked, video paging will be supported. If the caller sends a video page, the paging group members will be able to receive
Video Broadcast
and view the video.

Auto Record Enable this option to record in WAV format.

Allows the announcement to be played after the configured delay paging. If there are many messages, they will be played in
Delayed Paging
sequence.

LaggedTime(seconds) Set the delay paging duration, the default is 5 seconds.

Replace Display Name If enabled, the UCM will replace the caller display name with Paging/Intercom name.

Maximum Call
Specify the maximum call duration in seconds. The default value 0 means no limit.
Duration

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the

Custom Prompt users to upload the customized voice prompts.Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom
Prompt, where they could record new prompt or upload prompt files.

Members Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom
Select which extensions are allowed to use the paging/intercom feature for this paging group.
Whitelist
In case the user wants to broadcast a video, these requirements should be respected.

H.264 video encoding

.mkv or .tar/.tgz/tar.gz format

MKV files must be 30 MB file or less

Compressed files (.tar/.tgz/tar.gz) must be 50 MB or less.

File name can only contain alphanumeric characters, hyphens (-) and period (.)

If Auto Record is enabled, recorded video pages will be saved in MKV file format.Saved recordings can be found in the CDR🡪Recordings🡪Video
Recordings page.

Configure Announcement Paging

Announcement Paging

Enable Enable/Disable Announcement Paging.

Name Configure announcement group name.

Type Select “Announcement Paging”

Extension Extension

If checked, video paging will be supported. If the caller sends a video page, the paging group members will be able to receive
Video Broadcast
and view the video.
 Configure Announcement Paging transmission method.

Transmission Method Unicast: Depending on members selection.

Multicast: Depending on Multicast IP address and Port

Maximum Call
The maximum allowed duration of a call in seconds. Default value is 0 (no limit).
Duration (s)

Announcement File Configures an audio/video file to play to the paging members in this group.

Set the number of times to play the audio/video file in this announcement paging. To ensure the intended playback amount of the
Play Count
configured announcements, please set an appropriate maximum paging duration.

Repeat If enabled, the announcement page will be repeated for the selected weekdays.

Date Configure Announcement Paging Date.

Time Configure Announcement Paging Time.

Members Select available users from the left side to the paging/intercom group member list on the right.

Configure Private Intercom

Private Intercom feature allows initiating an intercom with the members of an intercom group. The members of the group will be able to hear the intercom

initiator, but they will not be able to hear each others’ voices. To configure this feature please navigate to the UCM web UI then go to Call Features →

Paging/Intercom then click on “Add”.

Name Configure paging/intercom group name.

Type Select “Private Intercom”.

Extension Configure the paging/intercom group extension.

Auto Record Enable this option to record in WAV format.

Replace Display
If enabled, the UCM will replace the caller display name with Paging/Intercom name.
Name

Maximum Call
Specify the maximum call duration in seconds. The default value 0 means no limit.
Duration

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users

Custom Prompt to upload the customized voice prompts. Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt,

where they could record new prompt or upload prompt files.

Members Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom
Select which extensions are allowed to use the paging/intercom feature for this paging group.
Whitelist

Private Intercom
Paging/Intercom Group Settings

Page/Intercom Group Settings

The UCM630X has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom

feature code, click on “Feature Codes” in the “Paging/Intercom Group Settings” dialog. Or users could go to Web GUI🡪Call Features🡪Feature Codes
directly.

Configure a Scheduled Paging/Intercom

Users can schedule paging/intercom calls by using the Schedule Paging/Intercom page. To schedule, click the Add button on the new page and configure the
caller, the group to use, and the time to call out.

Schedule Paging/Intercom page

Caller Configure the caller ID for the paging / intercom group.

Paging/Intercom
Select the paging / intercom group from the list of the available groups.
Group

Start Time Configure the start time of the scheduled paging / intercom call.

Select the type for the scheduled paging / intercom call. The available types are: Single time or Daily basis. Default is
Type
“Single”.

Action Status Display the action status of the scheduled paging / intercom call.

Table 86: Schedule Paging / Intercom Settings

 Creating a scheduled paging/intercom call

OPERATOR PANEL

Configure Operator Panel

Operator Panel settings can be accessed via Web GUI🡪Call Features🡪Operator Panel.

The UCM630X supports the operator panel so that UCM extension can be used as admin to manage calls and activities such as extension status, call queue
status, transfer, barge-in, hangup, etc. On Grandstream Wave client, it can display the extensions, ring group, voicemail, call queue, call park status under the

management of the extension. This section describes how to configure the operator panel.

Operator Panel Configuration Page

Click on “Add” to create the operator panel.

Click on to edit the operator panel.

Click on to delete the operator panel.

Name Configure the name for the operator panel created for identification purposes.

Assign the administrator for the operator panel. It can be an extension, a extension group or a department. For the selected extension
Administrator
groups and departments, subsequent extensions will automatically become administrators.

Management Module

The selected extensions will be supervised by the administrator, and you can choose extensions, extension groups, and departments.
Extension
For the selected extension groups and departments, subsequent extensions will be automatically supervised by the administrator.

The selected Ring Groups will be supervised by the administrator. If selecting “All”, all Ring Groups and subsequent updates will
Ring Groups
be automatically supervised by the administrator.

Voicemail Groups The selected Voicemail Groups will be supervised by the administrator. If selecting “All”, all Voicemail Groups and subsequent
 updates will be automatically supervised by the administrator.

The selected Call Queue will be supervised by the administrator. If selecting “All”, all Call Queue and subsequent updates will be
Call Queue
automatically supervised by the administrator.

The selected Parking Lot will be supervised by the administrator. If selecting “All”, all Parking Lot and subsequent updates will be
Parking Lot
automatically supervised by the administrator.

CALL QUEUE
The UCM630X supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming
calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUI🡪Call

Features🡪Call Queue.

Configure Call Queue

Call queue settings can be accessed via Web GUI🡪Call Features🡪Call Queue.

Call Queue

UCM630X supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller

could choose to leave a message/ transfer to default extension or keep waiting in the queue.

To configure this feature, please go to UCM Web GUI🡪Call Features🡪Call Queue🡪Create New Queue/Edit Queue🡪Queue Options🡪set Enable
Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.

Click on “Add” to add call queue.

Click on to edit the call queue. The call queue configuration parameters are listed in the table below.

Click on to delete the call queue.

Basic Settings

General

Extension Configure the call queue extension number.

Name Configure the call queue name to identify the call queue.

Strategy Select the strategy for the call queue.

● Ring All: Ring all available Agents simultaneously until one answers.
 ● Linear: Ring agents in the specified order.

● Least Recent: Ring the agent who has been called the least recently.

● Fewest Calls: Ring the agent with the fewest completed calls.

● Random: Ring a random agent.

● Round Robin: Ring the agents in Round Robin scheduling with memory.

The default setting is "Ring All".

Select the Music On Hold class for the call queue.

Music On Hold
Note: Music On Hold classes can be managed from Web GUI🡪PBX Settings🡪Music On Hold.

Configure the maximum number of calls to be queued at once. This number does not include calls that have been connected with

Max Queue Length agents, only calls that are still in queue. When this maximum value is exceeded, the caller will hear a busy tone and be forwarded

to the configured failover destination. Default value is 0 (unlimited).

Configure the amount of time in seconds after ending a call where the agent will not receive additional calls. Once this time has
Agent Rest Time (s)
passed, the agent will be able to receive calls again. If set to 0, agents can receive additional calls immediately after ending a call.

Retry Time (s) Configure the number of seconds to wait before ringing the next agent. The minimum is 1.

Agent Ring Time Configure the number of seconds to ring an agent. The minimum is 5.

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings
Auto Record
under Web GUI🡪Call Features🡪Call Queue.

Welcome Prompt

Enable Enable the welcome prompt.

Initial tone that plays when the user dials the queue number.
Custom Prompt
Note: The user can upload a custom prompt directly from this parameter.

Play Full Welcome If enabled, queue agents will not be rung until after the welcome prompt is done playing. Otherwise, queue agents will be rung

Prompt while the playing the welcome prompt.

Satisfaction Survey Prompt

After a queue agent hangs up a call, a prompt will play asking the caller to rate their satisfaction on a scale of 1 to 5, with 5 being

Custom Prompt the highest.

Note: The user can upload a custom prompt directly from this parameter.

Max Wait Time

Configures the amount of time a caller will be kept in queue before the the call is automatically routed to the configured Max
Max Wait Time
Wait Time Destination. If set to 0, callers will be kept in queue indefinitely.

Destination The call will be routed to this destination if no one in this queue answers the call.
Destination Prompt Cycle

Enable Enable Destination Prompt Cycle

Configure the voice prompt cycle (in seconds) of this call queue. When playing the voice prompt, you can press 1 to transfer to
Destination Prompt Cycle
failover destination.

When playing a custom prompt, press 1 to enter the failover destination or continue waiting in queue.
Custom Prompt
Note: The user can upload a custom prompt directly from this parameter.

After the specified amount of time, the caller will be prompted to press 1 to immediately get redirected to the configured failover
Destination
destination.

Advanced Settings

Virtual Queue

Enable Virtual Queue If enabled, system will enable a virtual queue for users waiting in queue.

When in DTMF mode, pressing 2 will manually trigger virtual queue. When in Timeout mode, virtual queue will automatically

be triggered when the configured Virtual Queue Period has passed. DTMF mode and Timeout mode require the caller to
Virtual Queue Mode
manually set a callback number. When in Auto mode, virtual queue will automatically be triggered when the configured Virtual

Queue Period has passed. The callback number will automatically be set to the caller's detected CID number.

Virtual Queue Period (s) The amount of time in seconds that must pass before virtual queue is offered to callers when using Timeout mode or Auto mode.

Virtual Queue Outbound

System will add this prefix to dialed numbers when calling back users.
Prefix

Enable Virtual Queue

If enabled, the system will inform callers waiting in the queue of their positions in line.
Position Announcement

Enable Virtual Queue

If enabled, the estimated wait time for the call to get answered will periodically be announced to the caller.
Wait Time Announcement

Enable Virtual Queue

If enabled, agents will have a set amount of time to answer a virtual queue callback.
Callback Timeout

Virtual Queue Welcome

Upload the file of your welcome prompt of the virtual queue.
Prompt

Announcement Settings

Enable Position
If enabled, the system will inform callers waiting in the queue of their positions in line.
Announcement

Enable Wait Time If enabled, the estimated wait time for the call to get answered will periodically be announced to the caller. Note: Wait time will
Announcement not be announced if less than one minute.

Announcement Interval The interval at which caller positions and estimated wait times will be announced.

Agent ID Announcement If enabled, a system prompt coontaining the agent ID will be played to the caller when answered by an agent.

Empty Queue

Configure whether the callers will be disconnected from the queue or not if the queue has no agent anymore. The default setting

is "Strict".

Leave When Empty ● Yes: Callers will be disconnected from the queue if all agents are paused or invalid.

● No: Never disconnect the callers from the queue when the queue is empty.

● Strict: Callers will be disconnected from the queue if all agents are paused, invalid or unavailable.

Configure whether the callers can dial into a call queue if the queue has no agent. The default setting is "No".

● Yes: Callers can always dial into a call queue.

Dial in Empty Queue
● No: Callers cannot dial into a queue if all agents are paused or invalid.

● Strict: Callers cannot dial into a queue if the agents are paused, invalid or unavailable.

Choose the destination where the call will be directed when the queue is empty or when all the agents are not logged in, here are

the destinations that can be configured:

● Play Sound.

● Extension.

● Voicemail.
Failover Destination
● Queues.

● Ring Group.

● Voicemail Group.

● IVR

● External Number.

CTI

Enable Agent Login Enabling agent login will cause the dynamic agents to be unavailable.

Queue Chairman The queue chairman can log into his web portal to operate the queue.

Service Level Agreement (SLA)

Toggles Service Level Agreement (SLA), which is percentage measurement of the queue group's ability to answer incoming calls

within a defined amount of time. If a queue group's calculated SLA percentage is below the configured threshold value, alerts will
Enable SLA
be generated and sent out via email to the specified recipients. Example: The SLA goal is 80% of calls (Threshold) within 20

seconds (SLA Time). If less than 80% of queue calls are answered within 20 seconds, the specified users will be notified of it.

Configures the amount of time in seconds that agents must answer incoming queue calls within to satisfy service quality

SLA Time (s) requirements. Answering calls past this time will negatively affect the SLA measurement, and an alert will be generated once it

hits below the specified SLA alert threshold. Supported values are 1 to 180. Default value is 20.
 SLA Alert Email
Enable SLA alert email notification.
Notification

Configures the SLA alert threshold. If the percentage of queue calls answered within the configured SLA Time go below this
Alert Threshold (%)
value, an alert email will be generated and sent to the configured recipients. Supported values are 1 to 100. Default value is 80.

Configures the minimum amount of time (in minutes) between alert sending. If a new alert is generated within this period, it will
SLA Alert Interval (m)
not be sent to recipients until the next alert interval. The valid range is from 1 to 120. The default value is 120.

SLA Alert Email

The template of the SLA alert email notifications.
Template

Send SLA alert notifications to the configured alert email recipients. If a recipient does not have an email address configured,
Alert Email Recipients
they will not receive the alert notifications.

Other Settings

If enabled, the UCM630X will report (to the agent) the duration of time of the call before the caller is connected to the agent. The
Report Hold Time
default setting is "No".

If enabled, the UCM will replace the caller display name with the Call Queue name so that the caller knows the call is incoming
Replace Display Name
from a Call Queue.

Enable Feature Codes Enable feature codes option for call queue. For example, *83 is used for “Agent Pause”

Autofill Configure to enable autofill.

Dynamic Login Password If enabled, the configured PIN number is required for dynamic agent to log in. The default setting is disabled.

Alert-Info When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.

Agents

Go to “Agents” Tab and Select the available users to be the static agents in the call queue. Choose from the available users on the

Static Agents left to the static agents list on the right. Click on ⮜ or ⮞ to choose. And use UP and Down arrow to select the order of the agent

within the call queue.

Number of agents limitation:

To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of agents allowed to be assigned depending on the UCM

model used. If the user attempts to configure the number of static agents to be more than the maximum allowed number, a warning message will appear.

The following table lists the maximum number of agents for each UCM model:

UCM Model Maximum Number of Agents in Call Queue

UCM6301 25
 UCM Model Maximum Number of Agents in Call Queue

UCM6302 50

UCM6304 80

UCM6308 160

Table 89: Number of Agent Limitation

Click on “Global Queue Settings” to configure Agent Login Extension Postfix and Agent Logout Extension Postfix. Once configured, users could log in the

call queue as dynamic agent.

Agent Login Settings

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to

login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent does not need to be listed as static agent and can log
in/log out at any time.

Call queue feature code “Agent Pause” and “Agent Unpause” can be configured under Web GUI🡪Call Features🡪Feature Codes. The default feature
code is *83 for “Agent Pause” and *84 for “Agent Unpause”.

Note: When dialing the “Agent Pause” feature code, users can specify the reason for it. The following reasons are available: (1) Lunch, (2) Hourly Break,

(3) Backoffice, (4) Email, and (5) Wrap.

The agent can also dial the feature with the number of the reason of the pause. E.g., if the agent want to perform a pause for lunch, he/she can dial *831

directly instead of waiting for the IVR response.

Queue recordings are shown on the Call Queue page under “Queue Recordings” Tab. Click on to download the recording file in .wav format; click on

to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected

Recording Files” or click on “Delete All Recording Files” to delete all recording files.

Call Center Settings and Enhancements

UCM supports light weight call center features including virtual queue and position announcement, allowing the callers to know their position on the call

queue and giving them the option to either stay on the line waiting for their turn or activate a callback which will be initiated by the UCM one an agent is free.

To configure call center features, press on an existing call queue and go under the advanced settings tab.

Following parameters are available:

Enable Virtual
Enable virtual queue to activate call center features.
Queue
Virtual Queue Configure the time in (s) after which the virtual queue will take effect and the menu will be presented to the caller to choose an
Period option. Default is 20s.

Offered to caller after timeout: After the virtual queue period passes, the caller will enter the virtual call queue and be presented

with a menu to choose an option, the choices are summarized below:

Press * to set current number as callback number.

Press 0 to set a callback number different than current caller number.

Press # to keep waiting on the call queue.

Virtual Queue

Mode
Triggered on user request: In this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with

the menu to choose an option as below:

Press * to set current number as callback number.

Press 0 to set a callback number different than current caller number.

Press # to keep waiting on the call queue.

Virtual Queue
System will add this prefix to dialed numbers when calling back users.
Outbound Prefix

When this option is enabled and after a caller registers a call back request on the virtual queue. While all the agents are busy, the
Enable Virtual
UCM will call an agent once he/she is idle again, this timeout is used for how long the UCM continues calling the agent and if the
Queue Timeout
agent doesn’t answer the call then the callback request will timeout and expire.

Write Timeout Configure the virtual queue callback timeout period in seconds.

Enable Virtual Enable the announcement of the caller’s position periodically.

Queue Position
Announcement Note: Queue position will now be announced to the caller upon entering the queue.

Position

Announcement Configure the period of time in (s) during which the UCM will announce the caller’s position in the call queue.

Interval

Enable Virtual

Queue Wait Time When enabled the UCM will announce the estimated queue wait time to callers if the estimated wait time is longer than 1 minute.

Announcement

Queue Chairman Select the extension to act as chairman of the queue (monitoring).

Virtual Queue
Click on “Upload Audio File” to upload the VQ welcome prompt.
Welcome Prompt
 When enabled, statics agents can conveniently log in and out of a queue by configuring a programmable key on their phones as a
shortcut.

Notes:

This feature is currently available only for GXP21xx phones on firmware 1.0.9.18 or greater.
Enable Agent

Login After enabling the feature, users need to set the option on GXP21XX phone under “Account🡪SIP Settings🡪Advanced

Features🡪Special Feature” to “UCM Call Center”. A softkey labeled “UCM-CC” will appear on the bottom of the phone’s
screen.

When this option is enabled, dynamic agent login will be no longer supported.

In case of concurrent registrations, changing agent status on one phone (login/logout) will be reflected on all phones.

Table 90: Call Center Parameters

Queue Auto fill enhancement:

The waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.

For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the

two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers

the call from the first caller.

Call Queue Statistics

Along with call center features, users can also gather detailed call queue statistics allowing them to make better changes/decision to manage better the call

distribution and handling based on time, agent, and queue.

To access call queue statistics, go to Web GUI🡪Call Features🡪Call Queue and click on “Call Queue Statistics”, the following page will be displayed:

Call Queue Statistics

Agent statistics: shows the number of calls and call-related information of agents;
 Queue Statistics: counts the number of calls in the queue and information such as calls, waiting, and callback;

Agent satisfaction statistics used for user’s rating of agents;

Queue satisfaction statistics counts the score survey statistics.

The overview page performs seat statistics, queue statistics, seat satisfaction statistics, and queue satisfaction statistics according to the business. Agent
statistics record the number of calls and call-related information of agents; queue counts the number of calls in the queue and information such as calls,

waiting, and callback; agent satisfaction statistics are survey statistics based on user ratings of agents; queue satisfaction statistics are user-queue The score

survey statistics.

By selecting a time interval, administrators can get detailed statistics for agent(s) such as total calls, answered calls etc, as well as for the queue(s) such as

ABANDONED CALLS also a detailed information for the queue’s call log by clicking on Options🡪Information button and the below window will pop up:

Queue’s call log details

User can download statistics on CSV format by clicking on the “Download”, also the statistics can be cleared using “Reset Statistics” button.

The statistics can be automatically sent to a specific email address on a preconfigured Period, this can be done by clicking on “Automatic Download”, and

user will be directed to below page where he can configure the download period (Day/Week/Month) and the Email where the statistics will be sent (Email

settings should be configured correctly):

Automatic Download Settings – Queue Statistics

Significantly more information is now available UCM’s queue statistics page. In addition to the information presented in previous firmware, users can now

view a call log that displays calls to all agents and queues, a dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be
obtained by pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV format and will be packaged into a
single tar.gz file upon download.

Agent Details is a call log that shows every call to each individual agent from all queues. The following information is available:

Time – the date and time the call was received.

Agent – the agent that was rung for the call.

Queue – the queue that the call went to.

Caller ID Number – the CID of the caller

Abandoned – indicates whether the call was picked up or not by that specific agent. If the call rang several agents simultaneously, and this specific agent

did not pick up the call, the call will be considered abandoned even if a different agent in the same queue picked it up.

Wait Time – the amount of time that the call was waiting in queue after dialing in.

Talk Time – the duration of the call after it was picked up by agent.

Agent details

Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the amount of time the dynamic agents were logged

in. Dynamic agents are extensions that log in and out either via agent login/logout codes (configured in Global Queue Settings page) or by using the GXP21xx
call queue softkey. A new record will be created only when an agent logs out. The following information is available:

Agent – the extension that logged in and out.

Queue – the queue that the extension logged in and out of.

Login Time – the time that the extension logged into the queue.

Logout Time – the time that the extension logged out of the queue.

Login Duration – the total length of time that the extension was logged in.

Login Record

Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time that agents are paused. If an agent is part of
several queues, an entry will be created for each queue. An entry will only be created after an agent unpauses. The following information is available:

Agent – the extension that paused and unpaused.

Queue – the queue that the agent is in.

 Pause Time – the time that the agent paused.

Resume Time – the time that the agent unpaused.

Pause Duration – the total length of time the agent was paused for.

Pause Log

Switchboard
Switchboard is a Web GUI tool for call queue monitoring and management, admin can access to it from the menu Call Features🡪Call Queue then press
“Switchboard”.

Following page will be displayed:

Switchboard Summary

Page above summarizes the available queues statistics and if one of the queues is clicked the user will be directed to page below:
 Call Queue Switchboard

The table below gives a brief description for the main menus:

Waiting This menu shows the current waiting calls along with the caller id and the option to hang-up call by pressing on the button.

Proceedi Shows the current established calls along with the caller id and the callee (agent) as well as the option to hang-up, transfer, add conference or

ng barge-in the call.

Displays the list of agents in the queue and the extension status (idle, ringing, in use or unavailable) along with some basic call statistics and

agent’s mode (static or dynamic).

Agents
Note: the dashboard will show the number of calls (answered and abandoned) of each agent. For dynamic agents, it will count the number of

calls starting from the last login time.

Table 91: Switchboard Parameters

There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue Chairman, and Queue Agent.

Super Admin – Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor, and execute actions for
incoming and ongoing calls for each extension in Switchboard, and generate Call Queue reports to track performance.

Queue Chairman – User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log
into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Other Features” in the side menu

and click on “Call Queue”. In the image below, User 1001 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents

of the queue in the Switchboard and their related information (Extension Status, Agent, Name, Answered, Abandoned, Login/Logout Time,

Pause/Resume Time, Talk Time, Agent Status, Pause Reason, and Options). The Chairman is also able to log out dynamic agents from call queues.
 Queue Chairman

Queue Agent – User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension
number and assigned user password. To access the Switchboard, click on “Other Features” in the side menu and click on “Call Queue”. However, a

queue agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension. In the image below, User 1000 is a queue
agent and can see only his own information in the Switchboard.

Queue Agent

Global Queue Settings

As explained before, under this section users can configure the feature codes for Dynamic agent login and logout, and also can now customize the keys for

virtual queue options like shown below.

 Global Queue Settings

Dynamic Agent Login Settings

Configure the code to dial after the queue extension to log into the queue (i.e. queue extension + suffix).
Agent Login Code Suffix
If no suffix is configured, dynamic agents will not be able to log in

Configure the code to dial after the queue extension to log out of the queue (i.e. queue extension + suffix).
Agent Logout Code Suffix
If no suffix is configured, dynamic agents will not be able to log out.

Virtual Queue Callback Key Settings

Enable Select whether to enable or disable virtual queue callback feature. By default it’s disabled.

Call Back Current Number Press the feature key configured to set your current number as callback number.

Custom Callback Number Press these feature key configured to set a custom callback number.

Continue Waiting Press the feature key configured to continue waiting.

Table 92: Global Queue Settings

PICKUP GROUPS
The UCM630X supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by

dialing “Pickup Extension” feature code (by default *8).

Configure Pickup Groups

Pickup Groups interface

Pickup groups can be configured via Web GUI🡪Call Features🡪Pickup Groups.

Click on to create a new pickup group.

Click the button to upload the pickup group information in CSV format.

Click the button to generate pickup group information in .CSV format.

Click on to edit the pickup group.

Click on to delete the pickup group.

Select extensions from the list on the left side to the right side.

Edit Pickup Group

Configure Pickup Feature Code

When picking up the call for the pickup group member, the user only needs to dial the pickup feature code. It is not necessary to add the extension number

after the pickup feature code. The pickup feature code is configurable under Web GUI🡪Call Features🡪Feature Codes.

The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows the ringing extension they can use **
followed by the extension number in order to perform the call pickup operation. The following figure shows where you can customize these features codes
 Edit Pickup Feature Code

MUSIC ON HOLD
Music On Hold settings can be accessed via Web GUI🡪PBX Settings🡪Music On Hold. In this page, users could configure music on hold class and upload
music files. The “default” Music On Hold class already has 5 audio files defined for users to use.

Music On Hold Default Class

Click on “Create New MOH Class” to add a new Music On Hold class.

Click on to configure the MOH class sort method to be “Alpha” or “Random” for the sound files.

Click on next to the selected Music On Hold class to delete this Music On Hold class.

Click on to start uploading. Users can upload:

Single files with 8KHz Mono Music file, or

Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can only be letters, digits, or special characters -_
 the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.

Users could also download all the music on hold files from UCM. In the Music On Hold page, click on

and the file will be downloaded to your local PC.

Click on to disable it from the selected Music On Hold Class.

Click on to enable it from the selected Music On Hold Class.

Select the sound files and click on to delete all selected Music On Hold files.

The UCM630X allows Users to select the Music On Hold file from WebGUI to play it. The UCM630X will initiate a call to the selected extension and play

this Music On Hold file once the call is answered.

Steps to play the Music On Hold file:

1. Click on the button for the Music On Hold file.

2. In the prompted window, select the extension to playback and click .

Play Custom Prompt

3. The selected extension will ring.

4. Answer the call to listen to the music playback.

Users could also record their own Music On Hold to override an existing custom prompt, this can be done by following those steps:

1. Click on .

2. A message of confirmation will pop up, as shown below.

Information Prompt

3. Click .

4. In the prompted window, select the extension to playback and click .

 Record Custom Prompt

5. Answer the call and start to record your new music on hold.

6. Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.

Once the MOH file is deleted, there are two ways to recover the music files.

Users could download the MOH file from this link: http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz

After downloading and unzip the pack, users could then upload the music files to UCM.

Factory reset could also recover the MOH file on the UCM.

BUSY CAMP-ON
The UCM630X supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party

becomes available given the previous attempted call has failed.

The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide:

https://documentation.grandstream.com/knowledge-base/busy-camp-on-2/

PRESENCE
UCM does support SIP presence feature which allows users to advertise their current availability status and willingness to receive calls, this way other users

can use their phones in order to monitor the presence status of each user and decide whether to call them or not based on their advertised availability.

This feature is different than BLF which is used to monitor the dialog status for each extension (Ringing, Idle or Busy). Instead the SIP presence module gives

more options for users to choose which state they want to put themselves in.

In order to configure the presence status of an extension from the web GUI, users can access the menu of configuration using one of the two following

methods:

From admin account, go under the menu Extension/Trunk🡪Extensions and choose the desired extension to edit then navigate to the “Features” tab.

OR

From the User Portal, go under the menu Basic Information🡪Extensions and navigate to the Features tab to have the following options.
 SIP Presence Configuration

Select which status to set from the presence status selection drop list, six options are available and below is a brief description of these states:

Available The contact is online and can participate in conversations/phone calls.

Away The contact is currently away (ex: for lunch break).

Chat The contact has limited conversation flexibility and can only be reached via chat.

Do Not Disturb The Contact is on DND (Do Not Disturb) mode.

Custom Presence Status Please enter the presence status for this mode on the Web GUI. Up to 64 characters.

Unavailable The contact is unreachable for the moment, please try to contact later.

Table 93: SIP Presence Status

Another option to set the presence status and which is more practical is using the feature code from the user’s phone, one the user dials the feature code

(default is *48), a prompt will be played to select which status they want to put themselves in, by pressing the corresponding key.

The feature code can be enabled and customized from the Web GUI🡪Call Features🡪Feature Codes.
 SIP Presence Feature Code

When a user does change his/her SIP presence status by making a call using presence feature code, the UCM will create a corresponding CDR entry showing
the call as Action type = PRSENCE_STATUS.

Presence Status CDR

FOLLOW ME
Follow Me is a feature on the UCM630X that allows users to direct calls to other phone numbers and have them ring all at once or one after the other. Calls
can be directed to users’ home phone, office phone, mobile and etc. The calls will get to the user no matter where they are. Follow Me option can be found

under extension settings page Web GUI🡪Extension/Trunk🡪Extensions.

To configure follow me:

1. Choose the extension and click on .

2. Go to the Follow me tab to add destination numbers and enable the feature.
 Edit Follow Me

3. Click on to add local extensions or external numbers to be called after ringing the extension selected in the first step.

4. Once created, it will be displayed on the follow me list. And you can click on to delete the Follow Me.

The following table shows the Follow Me configuration parameters:

Enable Configure to enable or disable Follow Me for this user.

If external number is added in the Follow Me, please make sure this option is enabled or the “Skip Trunk Auth” option of the
Skip Trunk Auth
extension is enabled, otherwise the external Follow Me number cannot be reached.

Music On Hold
Configure the Music On Hold class that the caller would hear while tracking users
Class

By default, it is enabled, and user will be asked to press 1 to accept the call or to press 2 to reject the call after answering a Follow Me
Confirm When call.
Answering
If it is disabled, the Follow Me call will be established once after the user answers.

Enable
When enabled, the call will be routed to the default destination if no one in the Follow Me extensions answers the call.
Destination
 Configure the destination if no one in the Follow Me extensions answers the call. The available options are:

Extension

Voicemail

Queues
Default

Destination
Ring Group

Voicemail Group

IVR

External Number

Follow Me The added numbers are listed here. Click on to arrange the order. Click on to delete the number. Click on

Numbers to add new numbers.

New Follow Me Add a new Follow Me number which could be a ‘Local Extension’ or ‘External Number’. The selected dial plan should have

Number permissions to dial the defined external number.

Dialing Order Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.

Table 94: Follow Me Settings

Click on “Follow Me Options” under Web GUI🡪Extension/Trunk🡪Extension page to enable or disable the options listed in the following table.

Playback Incoming Status Message If enabled, the PBX will playback the incoming status message before starting the Follow Me steps.

If enabled, the PBX will record the caller’s name from the phone so it can be announced to the callee in each
Record the Caller’s Name
step.

Playback Unreachable Status

If enabled, the PBX will playback the unreachable status message to the caller if the callee cannot be reached.
Message

Table 95: Follow Me Options

SPEED DIAL

Add Speed Dial

The UCM630X supports Speed Dial feature that allows users to call a certain destination by pressing one or four digits on the keypad. This creates a system-

wide speed dial access for all the extensions on the UCM630X.

To enable Speed Dial, on the UCM630X Web GUI, go to page Web GUI🡪Call Features🡪Speed Dial.

User should first click on . Then decide from one digit up to four digits combination used for Speed Dial and select a dial destination from

“Default Destination”. The supported destinations include extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group,

DISA, Dial by Name and external number.

 Speed Dial Destinations

List of Speed Dial

Import Speed Dial

The user can import speed dial entries from a csv file, this reduces the amount of configuring the same speed dial entries on different UCMs. To do this,

please click on “Import” as the figure below shows.

Import Dial Speed

Then select the csv file of the speed dial entries and click

Important

Please use UTF-8 encoding when importing a CSV file. CSV files can be opened using programs such as Notepad and saved as a UTF-8 encoded file.

Alert

Importing speed dial entries will overwrite the existing speed dials, if you wish to import new speed dial entries to the already existing ones, you will have to
export them then combine them together in one file before you import it.
 Note

The number of speed dial entries is limited to 100, therefore, the maximum number of entries you can import is 100. However, if the UCM had already more than

a 100 entries, the limit will be set to that specific number.

Export Speed Dial

To export speed dial entries, please click on export as the screenshot below shows, then choose the location where to save the csv file.

Export Dial Speed

DISA
In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near one of his extensions. However, he does have

access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be

able to call from the outside, whether it is using his cell phone, pay phone, regular PSTN, etc. After calling into UCM630X, the user can then dial out via the

SIP trunk or PSTN trunk connected to UCM630X as it is an internal extension.

The UCM630X supports DISA to be used in IVR or inbound route. Before using it, create new DISA under Web GUI🡪Call Features🡪DISA.

Click on to add a new DISA.

Click on to edit the DISA configuration.

Click on to delete the DISA.

Create New DISA

The following table details the parameters to set and configure DISA feature on UCM630X PBX.

Name Configure DISA name to identify the DISA.

 Configure the password (digit only) required for the user to enter before using DISA to dial out.
Password
Note: The password must be at least 4 digits.

Configure the permission level for DISA. The available permissions are "Internal", "Local", "National" and "International" from the

lowest level to the highest level.

Permission The default setting is "Internal". If the user tries to dial outbound calls after dialing into the DISA, the UCM630X will compared the

DISA's permission level with the outbound route's privilege level.

If the DISA's permission level is higher than (or equal to) the outbound route's privilege level, the call will be allowed to go through.

Configure the maximum amount of time the UCM630X will wait before hanging up if the user dials an incomplete or invalid number.
Response Timeout
The default setting is 10 seconds.

Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5
Digit Timeout
seconds.

If enabled, during an active call, users can enter the UCM630X Hangup feature code (by default it is *0) to disconnect the call or hang
Allow Hangup
up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is "No".

Replace Display
If enabled, the UCM will replace the caller display name with the DISA name.
Name

Once successfully created, users can configure the inbound route destination as “DISA” or IVR key event as “DISA”. When dialing into DISA, users will be
prompted with password first. After entering the correct password, a second dial tone will be heard for the users to dial out.

EMERGENCY

Emergency Calls
UCM supports configuration and management of numbers to be called in emergency situation, thus bypassing the regular outbound call routing process and

allowing users in critical situation to dial out for emergency help with the possibility to have redundant trunks as point of exit in case one of the lines is down.

UCM63xx series are fully compliant with Kari’s Law and Ray Baum’s Act, for more information, please refer to the following links:

https://www.fcc.gov/mlts-911-requirements

https://documentation.grandstream.com/knowledge-base/emergency-calls/

In addition, Emergency calls can be automatically recorded by toggling on the new Auto Record and recordings can be viewed in the new Emergency

Recordings tab on the same page. Additionally, users can have these recordings be sent to the configured email address(es).

Email alerts are also supported after enabling the notification for the event under “Maintenance 🡪 System Events”

To configure emergency numbers, users need to follow below steps:

1. Navigate on the web GUI under “Call Features 🡪 Emergency Calls”

2. Click on to add a new emergency number.

3. Configure the required fields “Name, Emergency Number and Trunk(s) to be used to reach the number”.

4. Save and apply the configuration.

 Emergency Number Configuration

The table below gives more description of the configuration Parameters when creating emergency numbers.

Configure the name of the emergency call.

Name
For example, “emergency911″,”emergency211” and etc.

Emergency Number Config the emergency service number. For example,”911″,”211″ and etc.

Emergency Level Select the emergency level of the number. Level “3” means the most urgent.

If this option is not enabled, when the lines of trunks which the coming emergency call routes by are completely occupied, the

Disable Hunt on Busy line-grabbing function will automatically cut off a line from all busy lines so that the coming emergency call can seize it for

dialing out. This option is not enabled by default.

This option sets a custom prompt to be used as an announcement to the person receiving an emergency call. The file can be
Custom Prompt
uploaded from the page “Custom Prompt”. Click “Prompt” to add additional record.

Use Trunks Select the trunks for the emergency call. Select one trunk at least and select five trunks at most.

Members Notified Select the members who will be notified when an emergency call occurs.
 Specify the number of digits that will be Stripped from the beginning of the dialed number before the call is placed via the
Strip
selected trunk.

Specify the digits to be Prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number
Prepend
is stripped.

Auto Record When enabled, emergency call will be automatically recorded.

Send Recording File When enabled recording files will be sent to the configured email address.

Email Address The email address to where the recording files will be sent.

911 Emergency Sample

Emergency Recordings
UCM6300 Series allows recording emergency calls and they can be found under WebUI → Call Feature → Emergency Calls → Emergency Recordings

Emergency Recordings

Emergency Location Mapping

In compliance with Kari’s Law and the Ray Baum’s Act, UCM’s Emergency Calls feature now supports

emergency location mapping. This will allow users to associate subnets with emergency location

identification numbers (ELINs), which can then be used by E911 service providers for example to determine
the location of callers. The new options can be found under Call Features→Emergency

Calls→Emergency Location Mapping.

 Emergency Location Mapping

ELIN: The emergency location identification number registered with the E911 provider. This number will be sent out as the emergency call’s CID

number.

Subnet: The network subnet that the ELIN will be associated with. The ELIN that is sent to E911 providers is based on the subnet that a calling endpoint

is registered from. Example: “xxx.xxx.xxx.xxx/24” or “xxxx:xxxx:xxxx:xxxx:xxxx:xxxx:xxxx:xxxx/64”.

Location: Location associated with the configured subnet. This is used for the UCM administrator’s reference.

Geolocation Routing: Toggles whether to include the Geolocation header in the emergency call SIP INVITE message. The Location field value will be

used as the Geolocation header value.

Important Note

Please note that ELIN Mapping is supported only on peer trunks. It would not apply on register trunks.

CALLBACK
Callback is designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The

callback feature provides an economic solution for reduce the cost from this.

The callback feature works as follows:

1. Configure a new callback on the UCM630X.

2. On the UCM630X, configure destination of the inbound route for analog trunk to callback.

3. Save and apply the settings.

4. The user calls the PSTN number of the UCM630X using the mobile phone, which goes to callback destination as specified in the inbound route.

5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.

6. The UCM630X will call back the user.

7. The user answers the call.

 8. The call will be sent to DISA or IVR which directs the user to dial the destination number.

9. The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the UCM630X instead of to the mobile phone directly. Therefore, the user will not be

charged on mobile phone services for long distance or international calls.

To configure callback on the UCM630X, go to Web GUI🡪Call Features🡪Callback page and click on . Configuration parameters
are listed in the following table.

Name Configure a name to identify the Callback. (Enter at least two characters)

Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the

CallerID Pattern CallerID match this pattern so that the caller can get callback after hanging up the call.

Note: If leaving as blank, all numbers are allowed to use this callback.

Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used
Outbound Prepend
to match the outbound route. ‘-’ is the connection character which will be ignored.

Delay Before Callback Configure the number of seconds to be delayed before calling back the user.

Configure the destination which the callback will direct the caller to. Two destinations are available:

● IVR
Destination
● DISA

The caller can then enter the desired number to dial out via UCM630X trunk.

BLF AND EVENT LIST

BLF
The UCM630X supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot. For example, on the user’s phone,

configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red,

meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.

On the Grandstream GXP series phones, the MPK supports “Call Park” mode, which can be used to park the call by configuring the MPK number as call park

feature code (e.g., 700). MPK “Call Park” mode can also be used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701).

Event List
Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same UCM630X and remote extensions on
the VOIP trunk can be monitored. The event list setting is under Web GUI🡪Call Features🡪Event List.

Click on “Add” to add a new event list.

Sort selected extensions manually in the Eventlist

Click on to edit the event list configuration.

 Click on to delete the event list.

Configure the name of this event list (for example, office_event_list). Please note the URI name cannot be the same as the extension
URI
name on the UCM630X. The valid characters are letters, digits, _ and -.

Local Extensions Select the available extensions/Extension Groups listed on the local UCM630X to be monitored in the event list.

If LDAP sync is enabled between the UCM630X and the peer UCM630X, the remote extensions will be listed under "Available
Remote Extensions
Extensions". If not, manually enter the remote extensions under "Special Extensions" field.

Special Extensions Manually enter the remote extensions in the peer/register trunk to be monitored in the event list. Valid format: 5000,5001,9000

Create New Event List

Remote extension monitoring works on the UCM630X via event list BLF, among Peer SIP trunks or Register SIP trunks (register to each other). Therefore,

please properly configure SIP trunks on the UCM630X first before using remote BLF feature. Please note the SIP end points need support event list BLF in
order to monitor remote extensions.

When an event list is created on the UCM630X and remote extensions are added to the list, the UCM630X will send out SIP SUBSCRIBE to the remote

UCM630X to obtain the remote extension status. When the SIP end points register and subscribe to the local UCM630X event list, it can obtain the remote

extension status from this event list. Once successfully configured, the event list page will show the status of total extension and subscribers for each event
list. Users can also select the event URI to check the monitored extension’s status and the subscribers’ details.

To configure LDAP sync, please go to UCM630X Web GUI🡪Extension/Trunk🡪VoIP Trunk. You will see “Sync LDAP Enable” option. Once enabled,

please configure password information for the remote peer UCM630X to connect to the local UCM630X. Additional information such as port number, LDAP
outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM630X and remote UCM630X need enable LDAP sync option with the same
password for successful connection and synchronization.

Currently LDAP sync feature only works between two UCM630Xs.

 (Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM630X PBX. However, it might not work the other way

around depending on whether the non-UCM630X PBX supports event list BLF or remote monitoring feature.

DIAL BY NAME
Dial by Name is a feature on the PBX that allows caller to search a person by first or last name via his/her phone’s keypad. The administrator can define the
Dial by Name directory including the desired extensions in the directory and the searching type by “first name” or “last name”. After dialing in, the PBX

IVR/Auto Attendant will guide the caller to spell the digits to find the person in the Dial by Name directory. This feature allows customers/clients to use the

guided automatic system to contact the enterprise employees without having to know the extension number, which brings convenience and improves business

image for the enterprise.

Dial by Name Configuration

The administrators can create the dial by name group under Web GUI🡪Call Features🡪Dial By Name.

Create Dial by Name Group

 Configure Extension First Name and Last Name

1. Name

Enter a Name to identify the Dial by Name group.

2. Extension

Configure the direct dial extension for the Dial By Name group.

3. Custom Prompt

This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the page “Custom Prompt”. Click “Upload Audio File”

to add additional record.

4. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached
by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under Web

GUI🡪Extension/Trunk🡪Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the

administrators can make sure unscreened calls will not reach the company employee if he/she does not want to receive them directly.

5. Prompt Wait Time

Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before
this wait time is reached. Otherwise, timeout will occur, and the call might hang up. The timeout range is between 3 and 60 seconds.

6. Query Type

Specify the query type. This defines how the caller will need to enter to search the directory.

By First Name: enter the first 3 digits of the first name to search the directory.

By Last Name: enter the first 3 digits of the last name to search the directory.

7. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out.

By Order: After the caller enters the digits, the IVR will announce the first matching party’s name and number. The caller can confirm and dial out if it is the
destination party, or press * to listen to the next matching result if it is not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call or press 9 for

results in next page.

The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the

destination list when configuring IVR and inbound route. If Dial by Name is set as a key pressing event for IVR, user could use ‘*’ to exit from Dial by Name,
then re-enter IVR and start a new event. The following example shows how to use this option.

Dial By Name Group In IVR Key Pressing Events

Dial by Name Group In Inbound Rule

Please refer to [Username Prompt Customization] for Username Prompt Customization.

ACTIVE CALLS AND MONITOR

The active calls on the UCM630X are displayed in Web GUI🡪System Status🡪Active Calls page. Users can monitor the status, hang up the call as well as
barge in the active calls in real time manner.

Active Calls Status

To view the status of active calls, navigate to Web GUI🡪System Status🡪Active Calls. The following figure shows extension 1004 is calling 1000. 1000 is
ringing.
 Status🡪PBX Status🡪Active Calls – Ringing

The following figure shows the call between 1000 and 5555 is established.

Status🡪PBX Status🡪Active Calls – Call Established

The gray color of the active call means the connection of call time is less than half an hour. It means this call is normal.

Call Connection less than half hour

The orange color of the active call means the connection of call time is greater than half an hour but less than one hour. It means this call is a bit long.

Call Connection between half an hour and one hour

The red color of the active call means the connection of call time is more than one hour. It means this call could be abnormal.
 Call Connection more than one hour

Hang Up Active Calls

To hang up an active call, click on icon in the active call dialog. Users can also click on to hang up all active calls.

Call Monitor
During an active call, click on icon and the monitor dialog will pop up.

Configure to Monitor an Active Call

In the “Monitor” dialog, configure the following to monitor an active call:

1. Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.

2. “Monitored Extension” must be one of the parties in the active call to be monitored.

3. Select spy mode. There are three options in “Spy Mode”.

Listen

In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the audio of the user on this extension will not be heard by
either party in the monitored active call.

Whisper

In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The user on this extension can only talk to the selected
monitored extension and he/she will not be heard by the other party in the active call. This can be usually used to supervise calls.

Barge

In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The call will be established similar to three-way conference.
 4. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s extension is required before the active call can be

monitored. This option can be used to avoid adding participant who has auto-answer configured, or call forwarded to voicemail.

5. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and start monitoring. If “Require Confirmation” is

enabled, the user will be asked to confirm to monitor the call.

Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to [Table 100: UCM630X Feature Codes] and

[Call Recording] section for instructions.

CALL FEATURES
The UCM630X supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the
UCM630X and describes how to use the call features.

Feature Codes

Feature Maps

- Default code: #1

- Enter the code during active call. After hearing "Transfer", you will hear dial tone. Enter the number to transfer to. Then the user will

be disconnected, and transfer is completed.

- Options:
Blind Transfer
● Disable

● Allow Caller: Enable the feature code on caller side only.

● Allow Callee: Enable the feature code on callee side only.

● Allow Both: Enable the feature code on both caller and callee.

- Default code: *2

- Enter the code during active call. After hearing "Transfer", you will hear the dial tone. Enter the number to transfer to and the user will

be connected to this number. Hang up the call to complete the attended transfer. In case of the called party does not answer, users could

press *0 to cancel the call and retrieve the first call leg.

- Options:
Attended Transfer

● Disable

● Allow Caller: Enable the feature code on caller side only.

● Allow Callee: Enable the feature code on callee side only.

● Allow Both: Enable the feature code on both caller and callee.

● Default code: *44 (Disabled by default).

● Seamless Transfer allows user to perform blind transfer using UCM feature code without having music on hold presented during the
Seamless Transfer transfer process, it minimizes the interruption during transfer, making the process smooth and simple.

● During an active call use the feature code (*44 by default) followed by the number you want to transfer to in order to perform the
seamless transfer.

Disconnect - Default code: *0

- Enter the code during active call. It will disconnect the call.

- Options:
 ● Disable

● Allow Caller: Enable the feature code on caller side only.

● Allow Callee: Enable the feature code on callee side only.

● Allow Both: Enable the feature code on both caller and callee.

- Default code: #72

- Enter the code during active call to park the call.

- Options:

Call Park ● Disable

● Allow Caller: Enable the feature code on caller side only.

● Allow Callee: Enable the feature code on callee side only.

● Allow Both: Enable the feature code on both caller and callee.

-Default code: *3

- Enter the code followed by # or SEND to start recording the audio call and the UCM630X will mix the streams natively on the fly as

the call is in progress.

- Options:
Start/Stop Call

Recording ● Disable

● Allow Caller: Enable the feature code on caller side only.

● Allow Callee: Enable the feature code on callee side only.

● Allow Both: Enable the feature code on both caller and callee.

Enable Recording
Enable the Recording Whitelist feature
Whitelist

Recording
Select extension in the whitelist that can use the *3 recording function.
Operation Whitelist

Feature Code
Set the maximum interval (ms) between digits for feature code activation
Digits Timeout

DND/Call Forward

Do Not Disturb
Default code: *77
(DND) Activate

Do Not Disturb
Default code: *78
(DND) Deactivate

Call Forward Busy - Default Code: *90

Activate - Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Busy

Default Code: *91
Deactivate

Call Forward No - Default Code: *92

Answer Activate - Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward No
Default Code: *93
Answer Deactivate

Call Forward
- Default Code: *72
Unconditional
- Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.
Activate

Call Forward

Unconditional Default Code: *73

Deactivate

Remote Call Enable this option and configure the Remote Call Forward Whitelist below to allow specific extensions to dial the remote call

Forward Enable forwarding feature codes to set call forwarding for any extension.

Remote DND / Call Forward Settings

Enable this option and configure the Whitelist below to allow specific extensions to dial feature codes to set DND or call forwarding for
Enable
any extension.

Remote Call

Forward Busy Configures and enables CFB for any extension.

Enable

Remote Call

Forward No Configures and enables CFNA for any extension.

Answer Enable

Remote Call

Forward Always Configures and enables CFU for any extension.

Enable

Remote DND
Enables Do Not Disturb for any extension.
Enable

Remote Call

Forward Busy Disables CFB for any extension.

Disable

Remote Call

Forward No Disables CFNA for any extension.

Answer Disable

Remote Call

Forward Always Disables CFU for any extension.

Disable
Remote DND
Disables Do Not Disturb for any extension.
Disable

Whitelist Extensions in this whitelist can configure DND or call forwarding for any extension via feature codes.

Feature Codes

Voicemail

Voicemail Access - Default Code: *98

Code - Enter *98 and follow the voice prompt. Or dial *98 followed by the extension and # to access the entered extension's voicemail box.

- Default Code: *97

My Voicemail
- Press *97 to access the voicemail box.

Voicemail Group
Dial this code to access group voicemail. If password is required, enter password followed by the pound (#) key.
Access Code

Direct Dial
Prefix used to dial directly to voicemail.
Voicemail Prefix

Call Queue

- Default Code: *83

Agent Pause
- Pause the agent in all call queues.

- Default Code: *84

Agent Unpause
- Unpause the agent in all call queues.

Dynamic Agent
Log the dynamic agent out of all queues.
Logout

Call Pickup

Picks up a ringing call for another extension.

Pickup on Ringing

Prefix
Example: If the prefix is **, and there is a call ringing ext 1008, dial **1008 from a different extension to pick up the call to 1008.

Picks up an ongoing call for another extension.

Pickup In-call
Example: If the feature code is *45, and ext 1008 is in a call, dialing *45 and then 1008 following the prompt will take that call.
Prefix
Note: The feature code user must be in the extension's Allowed to seamless transfer list to pick up calls for it.

Pickup Extension This is the feature code to pick up incoming calls for other extensions in the same pickup group. The default setting is *8.

Call Barging

Enable Spy Check this box to enable spy feature codes.

 This is the feature code to listen in on a call to monitor performance. Your line will be muted, and neither party will hear you. The
Listen Spy
default setting is *54.

Barge Spy This is the feature code to join in on the call to assist both parties. The default setting is *56.

This is the feature code to speak to only one party in the call. For example, you could whisper to employees to help them handle a call.
Whisper Spy
Only an employee on your account will be able to hear you. The default setting is *55.

PMS

PMS Wakeup
Dial this feature code to access PMS Wakeup Service. You can add, update, activate or deactivate PMS Wakeup Service.
Service

PMS Remote
Dial this code to add, update, activate, and deactivate PMS wakeup service for other extensions.
Wakeup Service

2 methods are available:

1. Dial the room status feature code + maid code, listen to the prompt and then the dial the appropriate key for the desired room status.

Update PMS Room Example: Maid with maid code 0001 dials *230001, listens to the room status options prompt, and then dials 1 to change room status to

Status Available.

2. Dial room status feature code*maid code*desired room status option key to quickly change the room status without needing to go

through the system voice prompts. Example: Maid with maid code 0001 dials *23*0001*1 to change room status Available.

Misc

Configure the paging prefix for paging. For example, if the Paging Prefix is set to *81, dial *816000 to initiate a paging call to extension
Paging Prefix
6000.

Configure the intercom prefix for intercom calls. For example, if the Intercom Prefix is set to *80, dial *806000 to initiate an intercom
Intercom Prefix
call to extension 6000.

Blacklist Add Follow the voice prompt to add a caller ID to blacklist.

Blacklist Remove Follow the voice prompt to remove a caller ID from blacklist.

Direct Dial Mobile If calling mobile phone numbers is permitted, use this prefix plus the extension number to dial the mobile phone number of this

Phone Prefix extension directly.

Call Completion If the caller wants to use CC to complete a call, he/she can dial this code. After the CC has been registered successfully, the system will

Request start to monitor the status of the callee. The system will call back the caller when the callee's extension is available.

Call Completion
If the caller has requested CC successfully, and he/she doesn't need to call back anymore, he/she can dial this code to cancel the request.
Cancel

Presence Status Dial this feature code to set the presence status of the extension.

Call Flip - Default code: *46

 - Dial this code to move the call of this extension from another device to the current device.

Wakeup Service Dial this feature code to access UCM Wakeup Service. You can add, update, activate or deactivate UCM Wakeup Service.

Whitelisted extensions will be able to use the Remote Extension Privilege Update feature code to remotely change any extension's

Remote Extension outgoing call privilege.

Privilege Update Note: After this function has been enabled, the extension is the whitelist can set the privilege for outgoing calls of any extension by

dialing the feature code.

Remote Extension Privilege Update Whitelist

Procedure:

Remote Extension 1. Dial *26 on the whitelisted extension, hear the prompt "Change extension's outgoing permission level, please enter the phone
Privilege Update number, then enter # key."
Whitelist 2. After the process, voice will prompt "Press 1 to set to internal, press 2 to set to local, press 3 to set to national, press 4 to set to
international."

3. After selecting, it will prompt "Change extension XXXX outgoing permission to XXX", and hang up.

The UCM630X also allows user to one click enable / disable specific feature code as shown below:

Enable/Disable Feature codes

Parking Lot
User can create parking lots and their related slots under Web GUI🡪 Call Features🡪 Parking Lot. In the Parking Lot page, users can create lots of their

own. This allows different groups within an organization to have their own parking lots instead of sharing one large parking lot with others. While creating a
new parking lot, users can assign it a range that they think is appropriate for the group that will use the parking lot.
 Parking Lot

User can create a new Parking lot by clicking on button “Add” :

New Parking Lot

Parking Lot Default Extension: 700

Extension During an active call, initiate blind transfer and then enter this code to park the call.

Parking Lot Name Set a name to the parking lot

Default Extension: 701-720

Parked Slots
These are the extensions where the calls will be parked, i.e., parking lots that the parked calls can be retrieved.

Use Parklot as If checked, the parking lot number can be used as extension. The user can transfer the call to the parking lot number to park

Extension the call. Please note this parking lot number range might conflict with extension range.

Default setting is 300 seconds, and the maximum limit is 99.999 seconds.

Parking Timeout (s) This is the timeout allowed for a call to be parked. After the timeout, if the call is not picked up, the extension who parks the

call will be called back.

Music On Hold
Select the Music on Hold Class.
Classes

Configures a callback failover destination when the extension that is called back is busy. The call will be routed to the destination
Failover Destination
number and this reduces the chance of dropping parked calls.

Ring All Callback on If enabled, all registered endpoints of the extension will ring when callback occurs. Otherwise, only the original endpoint will be

Timeout called back.

 Forward to
If enabled, the call will be routed to the configured destination upon timeout. Otherwise, the call will be routed back to the
destination on
original caller.
timeout

This option appears once Forward to Destination on Timeout is enabled. Upon park timeout, the call will be routed to the
Timeout Destination
configured destination.

Parking Lot Timeout

Adds an Alert-Info header to parking lot callbacks after the Parking Timeout has been reached.
Alert-Info

Table 101 : Parking Lot

Call Park
The UCM630X provides call park and call pickup features via feature code.

Park a Call

There are two feature codes that can be used to park the call.

Feature Maps🡪Call Park (Default code #72)

During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to 720) will be announced after parking the call.

Feature Misc🡪Call Park (Default code 700)

During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking lot number (default range 701 to 720) will be

announced after parking the call.

Retrieve Parked Call

To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked call is not retrieved after the timeout, the original

extension who parks the call will be called back.

Monitor Call Park CID Name Information (GXP21xx, GRP261x Phones Only)

Users can see the CID name information of parked calls. VPK/MPKs must be configured as “Monitored Call Park” with the desired parking lot extension.
The display will alternate between displaying the parking lot extension and the call’s CID name. There is no need to configure anything on the UCM.

Monitored Call Park CID name

Call Recording
The UCM630X allows users to record audio during the call. If “Auto Record” is turned on for an extension, ring group, call queue or trunk, the call will be

automatically recorded when there is established call with it. Otherwise, please follow the instructions below to manually record the call.

1. Make sure the feature code for “Start/Stop Call Recording” is configured and enabled.

2. After establishing the call, enter the “Start/Stop Call Recording” feature code (by default it is *3) followed by # or SEND to start recording.

3. To stop the recording, enter the “Start/Stop Call Recording” feature code (by default it is *3) followed by # or SEND again. Or the recording will be

stopped once the call hangs up.

4. The recording file can be retrieved under Web GUI🡪CDR. Click on to show and play the recording or click on to download the recording file.

Figure 229: Download Recording File from CDR Page

The above recorded call’s recording files are also listed under the UCM630X Web GUI🡪CDR🡪Recording Files.

Download Recording File from Recording Files Page

Enable Spy
If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for users to dial from any extension to perform the

corresponding actions.

Assume a call is on-going between extension A and extension B, user could dial the feature code from extension C to listen on their call (*54 by default),
whisper to one side (*55 by default), or barge into the call (*56 by default). Then the user will be asked to enter the number to call, which should be either

side of the active call, extension A or B in this example.

“Enable Spy” allows any user to listen to any call by feature codes. This may result in the leakage of user privacy.
Shared Call Appearance (SCA)
Shared Call Appearance (SCA) functionality has been added to the UCM. With SCA, users can assign multiple devices to one extension, configure endpoints

to monitor that extension, make actions on behalf of that extension such as viewing call status and placing and receiving calls, and even barging into existing

calls. To configure the SCA functionality, please follow the steps below:

1. Users can enable SCA by navigating to the Extensions page, editing the desired extension, and enabling the option SCA.

With SCA enabled, the Concurrent Registrations field can only have a value of 1.

Enabling SCA option under Extension’s Settings

2. After enabling the option, navigate to Call Features🡪SCA. The newly enabled SCA extension will be listed. Click the “+” button under the Options
column to add a number that will share the main extension’s call appearance, which will be called private numbers.

SCA Number Configuration

3. Configure the private number as desired.

 SCA Private Number Configuration

4. Once the private number has been created, users must now register a device to it. To properly register a device to the private number, use the configured

private number as the SIP User ID. Auth ID and Password will be the same as the main extensions. Once registration is complete, SCA is now
configured.

SCA Options

5. Next, configure the VPK or MPK to Shared for both the main extension and the private number. SCA is now configured for both endpoint devices.

The following table describe the SCA Number configuration setting:

Private Number Configures the private number for the SCA.

Related Shared Line Display the related shared line.

Whether enable this private number. If not enabled, this private number is only record in DB, it will not affect other
Enable This Number
system feature.

Allow Origination from This

Enable this option will allow calling from this private number. By default, it is enabled.
Number

Allow Termination to This

Enable this option will allows calls to this private number. By default, it is enabled.
Number

Table 102: Add SCA Private Number

The following table describes the options available when editing the SCA number:

Shared Line Number While SCA is enabled, this number will be the same as the extension number.

Allow Call Retrieve from

Allows remote call retrieval. Must be enabled in public hold. By default, it is enabled.
Another Location
 Alert All Appearances for Allows all SCA group members to ring when the SCA shared number is paged. If disabled, only the SCA shared
Group Paging Calls number will ring when paged. By default, it is disabled.

Multiple Call Arrangement Allows simultaneous calls in an SCA group. By default, it is disabled.

Allow Bridging between

Allows location bridging for SCA group. Must be enabled when using the Barge-In feature. By default, it is disabled.
Locations

Configures the notification in the bridge when another party join.

None: No notification sound.

Bridge Warning Tone Barge-In only: Notification sound will play when another party join.

Barge-In and Repeat: Notification sound will play when another party joins and repeat every 30 seconds.

By default, it is set to “Barge-In Only”.

Table 103: Editing the SCA Number

ANNOUNCEMENT
The Announcement feature (not to be confused with Announcement Paging and Announcement Center) is a feature that allows users to set an unskippable

audio file to play to callers before routing them to a configured destination. Announcements can be configured as a destination in the Inbound Routes page.

To configure Announcement, users need to follow below steps:

1. Navigate on the web GUI under “Call Features 🡪 Announcement”

2. Click on to add a new Announcement.

3. Configure the required fields Name, Prompt, Default Destination to be used for the announcement.

Save and apply the configuration.

Figure 235: Announcement settings

The table below gives more description of the configuration parameters when creating Announcement.

Name Configure the name of the Announcement.

Prompt Audio file that needs to be uploaded in order to be played for a specific destination.

Default Destination Select the destination where to play the audio file.

Table 104: Announcement Parameters

PBX SETTINGS
This section describes internal options that have not been mentioned in previous sections yet. The settings in this section can be applied globally to the
UCM630X, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUI🡪PBX

Settings🡪General Settings.

General Settings

General Preferences

Configure the global CallerID used for all outbound calls when no other CallerID is defined with higher priority. If
Global Outbound CID
no CallerID is defined for extension or trunk, the global outbound CID will be used as CallerID.

Configure the global CallerID Name used for all outbound calls. If configured, all outbound calls will have the
Global Outbound CID Name
CallerID Name set to this name. If not, the extension’s CallerID Name will be used.

Configure the number of seconds to ring an extension before the call goes to the user’s voicemail box. The default

setting is 60.
Ring Timeout
Note: This is the global value used for each extension if “Ring Timeout” field is left empty on the extension

configuration page.

Block calls for the configured duration. If Extensions->Features->Call Duration Limit and Outbound Routes->Call
Call Duration Limit
Duration Limit are not configured, General Settings->Call Duration Limit will be used.

If enabled, users will hear voice prompt before recording is started or stopped. For example, before recording, the
Record Prompt
UCM630X will play voice prompt “The call will be recorded”. The default setting is “No”.

Allow External Numbers to Cancel If enabled, external call parties will be given the option to decline the recording of calls. The IVR will prompt the

Recording user to dial *3 in order to cancel the call recording.

If enabled, the caller and callee's audio will be split into two channels during call recording. Not applicable to calls
Stereo Recording
with more than 2 parties.

Configure the audio channels for the calling party and the called party. If the caller is selected as the right channel,

Calling Channel the callee will be used for the left channel, and vice-versa.

Note: This option will be available when "Stereo Recording" is enabled.

International Call Prefix When this configuration is empty, International Call Prefix can be empty or +.

Extension Preferences

If enabled, a strong password policy will be enforced. This does not affect user login passwords, which must be
Enforce Strong Password
strong.

Enable Random Password If enabled, the extension will created with a randomly generated password.
 If enabled, an email will be sent to an extension's configured email address after creating it or modifying that
Send Extension Update Emails
extension's settings.

If set to “Yes”, users could disable the extension range pre-configured/configured on the UCM630X. The default

Disable Extension Range setting is “No”.Note: It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary

issues.

The default extension range assignment is:

● User Extensions: 1000-6299 User Extensions is referring to the extensions created under Web
GUI🡪Extension/Trunk🡪Extensions page.

● Pick Extensions: 4000-4999 This refers to the extensions that can be manually picked from end device when

being provisioned by the UCM630X. There are two related options in zero config page🡪Zero Config Settings,

“Pick Extension Segment” and “Enable Pick Extension”. If “Enable Pick Extension” under zero config settings
is selected, the extension list defined in “Pick Extension Segment” will be sent out to the device after receiving
the device’s request. This “Pick Extension Segment” should be a subset of the “Pick Extensions” range here.
This feature is for the GXP series phones that support selecting extension to be provisioned via phone’s LCD.
Extension Ranges
● Auto Provision Extensions: 5000-6299 This sets the range for “Zero Config Extension Segment” which is the
extensions can be assigned on the UCM630X to provision the end device.

● Meeting Extensions: 6300-6399 This extension range is used for creating meeting rooms.

● Ring Group Extensions: 6400-6499 This extension range is used for ring groups

● Queue Extensions: 6500-6599 This range of extensions is used for queueing

● Voicemail Group Extensions: 6600-6699 This extension range is used for voicemail groups.

● IVR Extensions: 7000-7100 This extension range is used for

● Dial By Name Extensions: 7101-7199 This extension range is used for Dial by Name feature

● FAX Extension: 7200-8200 This extension range is used for T.38 Fax.

Default Extension Segment Clicking this button will reset the extension range to their default values.

RTP Settings

RTP Settings

Table 106: Internal Options/RTP Settings

RTP Start Configure the RTP port starting number. The default setting is 10000.

RTP End Configure the RTP port ending address. The default setting is 20000.

Configure to enable or disable strict RTP protection. If enabled, RTP packets that do not come from the source of the RTP stream
Strict RTP
will be dropped. The default setting is "Disable".

RTP Checksums Configure to enable or disable RTP Checksums on RTP traffic. The default setting is "Disable".

ICE Support Configure whether to support ICE. The default setting is enabled.

ICE is the integrated use of STUN and TURN structure to provide reliable VoIP or video calls and media transmission, via a SIP

request/ response model or multiple candidate endpoints exchanging IP addresses and ports, such as private addresses and TURN
 server address.

Configure STUN server address. STUN protocol is a Client/Server and also a Request/Response protocol. It is used to check the

connectivity between the two terminals, such as maintaining a NAT binding entries keep-alive agreement. The default STUN

Server is stun.ipvideotalk.com.
STUN Server
Valid format:

[(hostname | IP-address) [':' port]

The default port number is 3478 if not specified.

BFCP UDP Start Configure BFCP UDP port starting number. The default setting is 50000.

BFCP UDP End Configure BFCP UDP port ending number. The default setting is 52999.

BFCP TCP Start Configure BFCP TCP port starting number. The default setting is 53000.

BFCP TCP End Configure BFCP TCP port ending number. The default setting is 55999.

Configure TURN server address. TURN is an enhanced version of the STUN protocol and is dedicated to the processing of
TURN Server
symmetric NAT problems.

TURN Server Name Configure turn server account name

TURN Server Password Configure turn server account password.

Connection Protocol Protocol used to connect to the TURN server.

Number of ICE This configures the number of pre-collected ICE candidates to gather and send to remote peers. The higher the number, the

Candidates greater the network traffic consumption.

Payload

The UCM630X payload type for audio codecs and video codes can be configured here.

AAL2-G.726 Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112.

DTMF Configured payload type for DTMF. The default setting is 101.

G.721 Compatible Configure to enable/disable G.721 compatible. The default setting is Yes.

G.726 Configure the payload type for G.726 if “G.721 Compatible” is disabled. The default setting is 111.

iLBC Configure the payload type for iLBC. The default setting is 97.

OPUS Configure the payload type for OPUS. The default setting is 123.

Audio FEC Payload Type Configure the Audio FEC Payload Type. The default setting is 127
 Audio RED Payload Type Configure the Audio RED Payload Type. Default setting is 122

H.264 Configure the payload type for H.264. The default setting is 99.

H.263P Configure the payload type for H.263+. The default setting is 100 103.

VP8 Configure the payload type for VP8. The default setting is 108.

Main Video FEC Configure the Main Video FEC

RTP FECC Configure the RTP FECC

RTX Configure the RTX

G.722.1 G.722.1：Low-complexity coder, 24kbps.

G.722.1C G.722.1C：Low-complexity coder, 48kbps.

Table 107: Internal Options/Payload

PBX Settings/Voice Prompt Customization

Record New Custom Prompt

In the UCM630X Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page, click on “Record” and follow the steps below to record new IVR
prompt.

Record New Custom Prompt

1. Specify the IVR file name.

2. Select the format (GSM or WAV) for the IVR prompt file to be recorded.

3. Select the extension to receive the call from the UCM630X to record the IVR prompt.

4. Click the “Record” button. A request will be sent to the UCM630X. The UCM630X will then call the extension for recording the IVR prompt from the

phone.

5. Pick up the call from the extension and start the recording following the voice prompt.
 6. The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play, or delete the recording.

Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on “Upload” in Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page to upload the file to

the UCM630X. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM630X:

PCM encoded.

16 bits.

8000Hz mono.

In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.

File size under 5M.

Upload Custom Prompt

Click on “choose file to upload” to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.

Download All Custom Prompt

On the UCM630X, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and

navigate to PBX Settings🡪Voice Prompt🡪Custom Prompt and click on ”Download All”. The following window will pop up in order to set a name for the
downloaded file.

Download All Custom Prompt

The downloaded file will have a .tar extension.

Call Failure Tone Settings

SIP Trunk Prompt Tone

Prompt Tone Settings tab has been added to the UCM to help users choose which prompt will be played by the UCM during call failure, the following voice
message responses have been added and can be set to be played for 4XX, 5XX, and 6XX call failures:

Default for 404 and 604 status codes: “Your call can’t be completed as dialed. Please check the number and dial again.”
 Default for 5xx status codes: “Server error. Please check your device.”

Default for 403 and 603 status codes: “The call was rejected by the server. Please try again later.”

Default for all other status codes: “All circuits are busy now. Please try again later.”

Additionally, custom voice messages recorded and uploaded in PBX Settings🡪Voice Prompt🡪Custom Prompt can be used for these failure responses
instead of the default messages.

SIP Trunk Prompt Tone

General Call Prompt Tones

Moreover, users also have the possibility to customize the prompt for typical call failure reasons like (no permission to allow outbound calls, busy lines,

incorrect number dialed …Etc.).

To customize these prompts user could record and upload their own files under “’PBX Settings 🡪 Voice Prompt 🡪 Custom Prompts” then select each one

for specific call failure case under “PBX Settings -> Call Prompt Tones 🡪 General Call Prompt Tones” page as shown on the following figure:
 General Call Propt Tones

File Manager
UCM supports automatic or manual recording of calls and storage of IM chat files. Only recording files and IM files can be stored locally or on the GDMS,

meanwhile, video recording files can only be stored on NAS. Local storage means that the files will be stored in the internal memory of the UCM. For extra

storage capacity the user can plug a USB flash drive or an SD card, the UCM will always store in the USB drive first, then the SD card. In case no storage
drive is attached, it will automatically start storing the files internally.

File Manager

If “Enable Auto Change” is selected, the files will be automatically saved in the available USB Disk or SD card plugged into the UCM630x. If both
USB Disk and SD card are plugged in, the files will be always saved in the USB Disk.
 When “Enable Auto Change” is enabled, the option “Storage Path Priority“will appear. It allows the user to configure the priority of each storage unit

in the priority list (The storage on top of the list has the highest priority). The default priority list is GDMS Cloud Storage > NAS > USB 1 > USB > SD
Card > Local

If “Local” is selected, the files will be stored in UCM630x internal storage. If a storage drive is inserted, the UCM63xx will store the files into the
storage drive instead of internal storage. Priority list is USB drive, then SD Card.

If “GDMS Cloud Storage” is selected, data will no longer be stored locally and if you need to listen to the recording, download the file to the computer

side and play it offline.

Once “USB Disk” or “SD Card” is selected, click on “OK”. The user will be prompted to confirm to copy the local files to the external storage device.

Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.

Recording Storage Category

Note

Once a storage device has filled up, the UCM will choose the next available storage device based on the Storage Path Priority.

On the UCM630X, users have the following options when select the categories to copy the files to the external device:

Recording Files: Copy the normal recording files to the external device.

Conference: Copy the conference recording files to the external device.

Queue: Copy the call queue recording files to the external device.

All: Copy all recording files to the external device.

NAS
The UCM supports adding and backing up recordings to a network-attached storage (NAS) server. Following table describes NAS settings:

Enable Enabled / Disable the NAS recording functionality.

 Configure the Domain or IP address of the NAS server.
Host
Note: Currently, only IP addresses are supported in the Host/IP field.

Share
Specify the name of the shared folder.
Name

Userna
Specify the account username to access the NAS server.
me

Passwor
Configure the account password to access the NAS server.
d

Security
Select a security mode based on the server settings to ensure proper connection establishment. The default value is ntlmssp.
Mode

If configured correctly, the Status field will show “Mounted”, and the newly added NAS server will be shown on the Mounted Netdisk List.

Status Additionally, the NAS will appear as a selectable storage option in the PBX Settings🡪Recording Storage page and CDR🡪Recording Files
page.

Table 108: NAS Settings

Note

If Network Storage Device has 1GB of storage space left, it will be considered unavailable the UCM will trigger the external disk usage alert.

SIP SETTINGS
The UCM630X SIP global settings can be accessed via Web GUI🡪PBX Settings🡪SIP Settings.

General

Realm For Digest Configure the hostname or domain name for the UCM630X. Realms MUST be globally unique according to RFC3261. The

Authentication default setting is Grandstream.

Bind UDP Port Configure the UDP port used for SIP. The default setting is 5060.

Bind IPv4 Address Configure the IPv4 address to bind to. The default setting is 0.0.0.0, which means binding to all addresses.

Bind IPv6 Address Configure the IPv6 address to bind to. The default is: “[::]” and it means to bind to all IP addresses.
 If enabled, the UCM630X allows unauthorized INVITE coming into the PBX and the call can be made. The default setting is
“No”.

Allow Guest Calls Warning:

Please be aware of the potential security risk when enabling “Allow Guest Calls” as this will allow any user with the UCM630X
address to dial into the UCM630X.

If set to “No”, all transfers initiated by the endpoint in the UCM630X will be disabled (unless enabled in peers or users). The
Allow Transfer
default setting is “Yes”.

When sending MWI NOTIFY requests, this value will be used in the “From:” header as the “name” field. If no “From User” is
MWI From
configured, the “user” field of the URI in the “From:” header will be filled with this value.

Enable Diversion
If disabled, the UCM will not forward the diversion header.
Header

If enabled, collect calls will be blocked.

Block Collect Calls

Note: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-

Service-Info: service-code=Collect Call”.

Table 109: SIP Settings/General

MISC

Outbound SIP Registrations

Register Timeout Configure the register retry timeout (in seconds). The default setting is 20.

Configure the number of registration attempts before the UCM630X gives up. The default setting is 0, which means the UCM630X will
Register Attempts
keep trying until the server side accepts the registration request.

Trunk Register Configures the time window within which to send initial trunk registration requests. Instead of sending out all initial trunk reqistration

Period (s) requests at once, requests will be randomly sent out within this period.

Video

Max Bit Rate

Configure the maximum bit rate (in kb/s) for video calls. The default setting is 384.
(kb/s)

Support SIP
Select to enable video support in SIP calls. The default setting is “Yes”.
Video

Security
 Reject Non- If enabled, when rejecting an incoming INVITE or REGISTER request, the UCM630X will always reject with “401 Unauthorized”

Matching instead of notifying the requester whether there is a matching user or peer for the request. This reduces the ability of an attacker to scan

INVITE for valid SIP usernames. Default setting is “No”.

SDP Attribute Passthrough

Enable Attribute
If enable, and if the service does not know the attribute of FEC/FECC/BFCP, then the attribute will be passthrough.
Passthrough

Early Media

Enable Use Final

If enabled, call negotiation will use final response SDP.
SDP

Ignore 180
If enabled, ringing indication after 183 response will be ignored.
Response

Blind Transfer

Allow callback
If enabled, the UCM will call back to the transferrer when blind transfer fails (reason of failure includes busy and no answer).
when blind
Note: This feature takes effect only on internal calls.
transfer fails

Blind transfer
Configure the timeout in (s) for the transferrer waiting for the destination to answer. Default is 60s.
timeout

Hold

Configure the UCM to forward HOLD requests instead of processing holds internally. This serves to meet the standards set by some
Forward HOLD
providers that require HOLD requests to be passed along from endpoint to endpoint. This option is disabled by default.
Requests
Note: Enabling this option may cause hold retrieval issues and MOH to not be heard.

Session Timer

Force Timer If checked, always request, and run session timer.

Timer If checked, run session timer only when requested by other UA.

Session Expire Configure the maximum session refresh interval (in seconds). Default is 1800.

Configure the minimum session refresh interval (in seconds).

Min SE
The default setting is 90.

Table 111: SIP Settings/Session Timer

TCP & TLS

TCP Enable Configure to allow incoming TCP connections with the UCM630X. The default setting is “No”.

Configure the IP address for the TCP server to bind to. “0.0.0.0” means binding to all interfaces. The port number is optional, and
TCP Bind IPv4 Address
the default port number is 5060. For example, 192.168.1.1:5062.

Configure the IPv6 address for the TCP server to bind to. “[::]” means bind to all interfaces. The port number is optional with the
TCP Bind IPv6 Address
default being 5060. For example, [2001:0DB8:0000:0000:0000:0000:1428:0000]:5060.

TLS Enable Configure to allow incoming TLS connections with the UCM630X. The default setting is “Yes”.

Configure the IPv4 address for TLS server to bind to. “0.0.0.0” means binding to all interfaces. The port number is optional, and

TLS Bind IPv4 Address the default port number is 5061. For example, 192.168.1.1:5063. Note: The IP address must match the common name (host name)

in the certificate so that the TLS socket will not bind to multiple IP addresses.

Configure the IPv6 address for TLS server to bind to. “[::]” means bind to all interfaces. The port number is optional with default

TLS Bind IPv6 Address being 5061. For example, [2001:0DB8:0000:0000:0000:0000:1428:0000]:5061. Note: The IP address must match the common

name (host name) in the certificate so that the TLS socket will not bind to multiple IP addresses.

TLS Do Not Verify If enabled, the TLS server’s certificate will not be verified when acting as a client. The default setting is “Yes”.

This is the CA certificate if the TLS server being connected to requires self-signed certificate, including server’s public key. This
TLS Self-Signed CA
file will be renamed as “TLS.ca” automatically.

Reset Certificates Clicking on this button will reset the certificates.

Private Certificate and Key

This is the Certificate file (*.pem format only) used for TLS connections. It contains private key for client and signed certificate

TLS Cert for the server. This file will be renamed as “TLS.pem” automatically.

Note: The size of the uploaded certificate file must be under 2MB.

This file must be named with the CA subject name hash value. It contains CA’s (Certificate Authority) public key, which is used to

TLS Key verify the accessed servers.

Note: The size of the uploaded CA certificate file must be under 2MB.

Reset Certificates Clicking on this button will reset the certificates.

Cipher Suite

By default, all SIP TLS encryption suites are in effect on the system, and when turned on, you can configure the encryption suites
Restrict Cipher List
allowed to be used.

Cipher Suite Select the encryption suites that are allowed to be used for SIP TLS connections, in the order of priority as configured.
NAT

Configure a static IP address and port (optional) used in outbound SIP messages if the UCM630X is behind NAT. If it is a host name,
External Host
it will only be looked up once.

Use IP address in
If enabled, the SDP connection will use the IP address resolved from the external host.
SDP

External UDP
Configure externally mapped UDP port when the PBX is behind a static NAT or PAT.
Port

External TCP
Configure the externally mapped TCP port when the UCM630X is behind a static NAT or PAT.
Port

External TLS
Configures the externally mapped TLS port when UCM630X is behind a static NAT or PAT.
Port

Specify a list of network addresses that are considered inside of the NAT network. Multiple entries are allowed. If not configured, the
external IP address will not be set correctly.
Local Network

Address A sample configuration could be as follows:

192.168.0.0/16

Table 113: SIP Settings/NAT

ToS

ToS for SIP Configure the Type of Service for SIP packets. The default setting is None.

ToS for RTP Audio Configure the Type of Service for RTP audio packets. The default setting is None.

ToS for RTP Video Configure the Type of Service for RTP video packets. The default setting is None.

Default Configure the default duration (in seconds) of incoming/outgoing registration.

Incoming/Outgoing

Registration Time The default setting is 120.

Max
Configure the maximum duration (in seconds) of incoming registration and subscription allowed by the UCM630X. The default
Registration/Subscri
setting is 3600.
ption Time

Min
Configure the minimum duration (in seconds) of incoming registration and subscription allowed by the UCM630X. The default
Registration/Subscri
setting is 60.
ption Time
Enable Relaxed
Select to enable relaxed DTMF handling. The default setting is “No”.
DTMF

Select DTMF mode to send DTMF. The default setting is RFC4733. If “Info” is selected, SIP INFO message will be used. If

DTMF Mode “Inband” is selected, a-law or u-law are required. When “Auto” is selected, “RFC4733” will be used if offered, otherwise “Inband”

will be used. The default setting is “RFC4733”.

During an active call, if there is no RTP activity within the timeout (in seconds), the call will be terminated. The default setting is
no timeout.
RTP Timeout

Note: This setting does not apply to calls on hold.

When the call is on hold, if there is no RTP activity within the timeout (in seconds), the call will be terminated. This value of RTP
RTP Hold Timeout
Hold Timeout should be larger than RTP Timeout. The default setting is no timeout.

This feature can be used to avoid abnormal call drop when the remote provider requires RTP traffic during proceeding.

For example, when the call goes into voicemail and there is no RTP traffic sent out from UCM, configuring this option can avoid

RTP Keep-alive voicemail drop.

When configured, RTP keep-alive packet will be sent to remote party at the configured interval. If set to 0, RTP keep-alive is

disabled.

100rel Configure the 100rel setting on UCM630X. The default setting is “Yes”.

Trust Remote Party

Configure whether the Remote-Party-ID should be trusted. The default setting is “No”.
ID

Send Remote Party

Configure whether the Remote-Party-ID should be sent or not. The default setting is “No”.
ID

Configure whether the UCM630X should generate Inband ringing or not. The default setting is “Never”.

Yes: The UCM630X will send 180 Ringing followed by 183 Session Progress and in-band audio.
Generate In-Band
No: The UCM630X will send 180 Ringing if 183 Session Progress has not been sent yet. If audio path is established already
Ringing
with 183 then send in-band ringing.

Never: Whenever ringing occurs, the UCM630X will send 180 Ringing as long as 200OK has not been set yet. Inband
ringing will not be generated even the end point device is not working properly.

Server User Agent Configure the user agent string for the UCM630X.

Send Compact SIP

If enabled, compact SIP headers will be sent. The default setting is “No”.
Headers

Passthrough PAI
Passthrough PAI Header
Header
 Table 114: SIP Settings/ToS

STIR/SHAKEN
To prevent robocalls, UCM now supports STIR/SHAKE protocols. Related options have been added as a new tab in the SIP Settings page.

Clicking on the Add button will show the following window:

STIR/SHAKEN

STIR/SHAKEN – Add Authorized CID and Attestation level

Authorized CID Configure the authorized CID.

Configure the attestation level, which is the level of confidence of the carrier that the CID has not been spoofed. The following options

are available:

● A (Full attestation): The carrier is associated with the caller and the number. There is high confidence that the CID has not been
spoofed.
Attestation
● B (Partial attestation): The carrier is associated with the caller but not the number. There is uncertainty about whether the CID has
been spoofed or not.

● C (Gateway attestation): The carrier is not associated with the caller and has no confidence at all about the number. Generally
used for traceback.

Clicking on the Certificate Settings button will bring up the following window:
 STIR/SHAKEN – Certificate Settings

Certificate Download
Configure the public key download timeout period, the default value is 2 seconds.
Time (s)

Signature Valid Time

Configure the validity period of the digital signature, the default value is 15 seconds.
(s)

Configure the Private key.

Private Key
Note: The uploaded file must be less than 2MB in file size, only supports the .key format and must be ECC type. This file will

automatically be renamed to “private.key”.

Configure the Public Key.

Public Key
Note: The uploaded file must be less than 2MB in file size, only supports the .crt format and must be ECC type. This file will

automatically be renamed to “public.crt”.

Table 116: SIP Settings/STIR/SHAKEN – Certificate Settings

Transparent Call-Info header

UCM supports transparent call info header in order to integrate GDS door system with GXP21XX/GRP261X phones, the UCM will forward the call-info

header to the phone in order to request the live view from GDS door system and give the option to open the door via softkey.

Transparent Call-Info

IAX SETTINGS
The UCM630X IAX global settings can be accessed via Web GUI🡪PBX Settings🡪IAX Settings.

General

Bind Port Configure the port number that the IAX2 will be allowed to listen to. The default setting is 4569.

Bind IPv4
Force IAX2 to bind to a specific address instead of all addresses.
Address

Bind IPv6
Configure the IPv6 address to bind to. “[::]” means to bind to all IP addresses.
address

IAX1
Select to configure IAX1 compatibility. The default setting is “No”.
Compatibility

No If selected, UDP checksums will be disabled and no checksums will be calculated/checked on systems supporting this feature. The default
Checksums setting is “No”.

Delay Reject If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The default setting is “No”.

ADSI Select to enable ADSI phone compatibility. The default setting is “No”.

Music On
Specify which Music On Hold class this channel would like to listen to when being put on hold. This music class is only effective if this
Hold
channel has no music class configured and the bridged channel putting the call on hold has no “Music On Hold Suggest” setting.
Interpret

Music On
Specify which Music On Hold class to suggest to the bridged channel when putting the call on hold.
Hold Suggest

Bandwidth Configure the bandwidth for IAX settings. The default setting is “Low”.

Table 117: IAX Settings/General

Registration

IAX Registration

Options

Min Reg Expire Configure the minimum period (in seconds) of registration. The default setting is 60.

Max Reg Expire Configure the maximum period (in seconds) of registration. The default setting is 3600.

IAX Thread Count Configure the number of IAX helper threads. The default setting is 10.

IAX Max Thread

Configure the maximum number of IAX threads allowed. The default is 100.
Count
Auto Kill If enabled and no ACK is received for new messages after the specified wait time, the connection will be terminated.

Authentication
If enabled, authentication traffic in debugging will not show. The default is “No”.
Debugging

Configure codec negotiation priority. The default setting is “Reqonly”.

Caller

Consider the callers preferred order ahead of the host’s.

Host

Consider the host’s preferred order ahead of the caller’s.

Codec Priority

Disabled

Disable the consideration of codec preference all together.

Reqonly

This is the same as “Disabled”, except when the requested format is not available. The call will only be accepted if the requested

format is available.

Type of Service Configure ToS bit for preferred IP routing.

IAX Trunk Options

Trunk Frequency Configure the frequency of trunk frames (in milliseconds). The default is 20.

Trunk Time Stamps If enabled, time stamps will be attached to trunk frames. The default is “No”.

Table 118: IAX Settings/Registration

Security

Enter a single IP address (e.g., 1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) for which call token
Call Token Optional
validation is not required.

Max Call Numbers Configure the maximum number of calls allowed for a single IP address.

Max Unvalidated Call

Configure the maximum number of Unvalidated calls for all IP addresses.
Numbers

Max Call Numbers Configure to limit the number of calls for a give IP address of IP range.

IP or IP Range Enter the IP address (1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) to be considered for call number limits.

Table 119: IAX Settings/Static Defense

INTERFACE SETTINGS

Analog Hardware
The analog hardware (FXS port and FXO port) on the UCM630X will be listed in this page. Click on to edit signaling preference for FXS port or

configure ACIM settings for FXO port.

Select “Loop Start” or “Kewl Start” for each FXS port. And then click on “Update” to save the change.

FXS Ports Signaling Preference

For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port. Or users could click on “Detect” and

choose the detection algorithm, two algorithms exist (ERL, Pr) for the UCM630X to automatically detect the ACIM value. The detecting value will be

automatically filled into the settings.

FXO Ports ACIM Settings

Select country to set the default tones for dial tone, busy tone, ring tone and etc. to be sent from the FXS port. The default setting is
Tone Region
“United States of America (USA)”.

Advanced

Settings

Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment,
FXO
Minimum Operational Loop Current, and AC Impedance as predefined for your country’s analog line characteristics. The default setting is
Opermode
“United States of America (USA)”.

Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment,
FXS
Minimum Operational Loop Current, and AC Impedance as predefined for your country’s analog line characteristics. The default setting is
Opermode
“United States of America (USA)”.

Configure to enable or disable override Two-Wire Impedance Synthesis (TISS). The default setting is No.
FXS TISS

Override
If enabled, users can select the impedance value for Two-Wire Impedance Synthesis (TISS) override. The default setting is 600Ω.
 Select the codec to be used for analog lines. North American users should choose PCMU. All other countries, unless already known,
should be assumed to be PCMA. The default setting is PCMU.
PCMA

Override Note:

This option requires system reboot to take effect.

Configure whether normal ringing voltage (40V) or maximum ringing voltage (89V) for analog phones attached to the FXS port is
Boost Ringer
required. The default setting is “Normal”.

Fast Ringer Configure to increase the ringing speed to 25HZ. This option can be used with “Low Power” option. The default setting is “Normal”.

Configure the peak voltage up to 50V during “Fast Ringer” operation. This option is used with “Fast Ringer”. The default setting is
Low Power
“Normal”.

If set to “Full Wave”, false ring detection will be prevented for lines where Caller ID is sent before the first ring and proceeded by a
Ring Detect
polarity reversal, as in UK. The default setting is “Standard”.

Configure the type of Message Waiting Indicator on FXS lines. The default setting is “FSK”.

FXS MWI
FSK: Frequency Shift Key Indicator
Mode

NEON: Light Neon Bulb Indicator.

FXO

Frequency Allows users to adjust the tolerance of the FXO ringing frequency. 63Hz is considered the standard value and is selected by default.

Tolerance

Table 120: PBX Interface Settings

DAHDI Settings
When users encounter issues such as audio delay in outbound calls using the analog trunk, they can adjust DAHDI settings on the UCM to attempt to lessen or

resolve the issues.

DAHDI Settings

For the value of the option such as “32, half”:

The number in the option indicates the number of read/write buffers for TDM (DAHDI).

The “Half”, “Immediate” or “Full” option indicates the strategy when reading/writing data from buffer.
 “Half”: Data will be read/written from buffer when half of the buffer is occupied with data.

“Immediate”: Read/write from buffer whenever there is data occupying the buffer.

“Full”: Data will be read/written from buffer when buffer is fully occupied with data.

Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog trunk/Fax-related issues.

CONTACTS
Address book management is under UCM web UI->Maintenance, and it has two sections “Contact Management” and “Department management”.

Contact Management
Contact management page displays extension contacts and external contacts information.

Extension contacts

Extension contacts page shows all the extensions that has “Sync Contact” option enabled in extension settings page. The extension contacts here can be edited

or deleted individually or in batch. No new extension contact can be added directly from this page. If an extension contact is deleted from this page, “Sync

Contact” option is disabled from this extension. This will not delete the extension from UCM.

Note

“Delete” extension contact will only remove this extension from extension contact page and it will not sync to contacts on UCM. The extension itself still exists on

UCM.
 Extension Contacts

Note

“Delete” extension contact will only remove this extension from extension contact page and it will not sync to contacts on UCM. The extension itself still exists on
UCM.

Click Edit icon to configure name, department, email address and etc for each extension contact.
 Edit Extension Contact

Extension Displays extension number.

First Name Configure first name for the extension contact.

Last Name Configure last name for the extension contact.

Department Select department for the extension contact. Department can be created in “Department Management” page.

Department Title Configure the job title for the extension contact.

Email Address Configure email address for the extension contact.

Mobile Phone Number Configure mobile phone number for the extension contact.

Home Number Configure home number for the extension contact

Fax Configure Fax for the extension contact.

Same as Department
When this option is enabled, the contact extension will inherit the same privilege as the department it belongs to.
Contact Privileges

This option allows configuring privileges for the contact extension.

Contact View Privileges
Note: This option will be disabled if "Same as Department Contact Privileges" has been enabled.
 External contacts

On external contacts page, the admin can create single external contact, import contacts in batch, edit contacts, delete contacts and export contacts.

External Contacts

Click on “Export” icon, a CSV format file will be generated with the current external contacts.

Click on “import” icon, then follow the steps below to add external contacts in batch:

Step 1: For option “On Duplicate External Contacts”, select whether to skip duplicate contact on the imported CSV file or update the duplicate UCM

contact with the information in the CSV.

Step 2: Choose file from local PC to upload.

Step 3: Click on “Upload”.

Step 4: Click on “Apply” to complete importing external contacts.

Import External Contacts

Department Management
Departments are organizational units that allows organizing extensions within groups that specify the specialty of a the extension owners within a company.
This makes finding contacts easier within the UCM contact books.
 Department Management

Click on “Add” to create a new department. Configure the department name and select the superior department. By default the superior department is the root

directory. If the UCM has cloud IM configured, the root directory will be the deparment in cloud IM.

On the department list:

Click on to create sub department.

Click on to add member to the department.

Click on to edit the department.

Edit Department

Department Name Enter the name of the department.

Upper Level Department Select the upper level department if the department being created is a nested department.

Contact View Privileges

● All Contacts: The extensions in this department will be able to see all the contacts.
 ● Department & Sub-department Contacts: The extensions in this department will only be able to see the contacts which
are in the same department or in sub-departments.

Enable this option to share this department across the UCMs which use the same Cloud IM server. To be able to enable this
Set as Shared Department
option, make sure that the UCM has a RemoteConnect Plan and is correctly connected to the Cloud IM server.

Share to Following Sites Pick the sites to which you want to share this specific department.

Note

The user can create up to 100 departments with up to 4 levels of nested departments.

Important

To be able to use shared department, the UCM devices will have to be subscribed to a RemoteConnect plan that offers Cloud IM service. For more information

please refer to: https://ucmrc.gdms.cloud/plans

Privilege Management
The user can configure custom privileges other than the default ones (All contacts, Departments and sub-departments contacts). These custom privileges allow

more flexible ways of allowing contacts to view all or specific contacts from other departments.

UCM admin can add or edit Privilege Management; under UCM web UI🡪Contacts Privilege Management, there are 2 default privileges:

Visible to all contacts.

Only the contact person’s department and sub-department contacts are visible.

When Cloud IM is enabled on the UCM, a third privilege becomes available to choose:

Local Contacts: Restricts the contacts shown to the contacts of the local UCM.

Privilege Management — Cloud IM Disabled

DEVICE MANAGEMENT

IPC Devices
The UCM admin can add IPC devices and edit accessible extensions so these extensions can view the surveillance streams for the IPC devices.

Click on “Add” to add IPC device.

 IPC Devices

Edit the IPC device settings in the table below.

Device Number The number that allowed members can dial to access the IP camera.

Device Name Enter the name that you want to allocate for the device.

Protocol The media protocol that will be used.

IP Address Enter the IP address of the IP camera.

Port Enter the port of the IP camera. The default is 554

Channel Path If you want to view the stream of the specified channel, please configure the path of this stream.

Username If a username and password are set on this device, fill in this field to allow the UCM to access the device.

Password If a username and password are set on this device, fill in this field to allow the UCM to access the device.

Transmission Protocol Transport protocol of the IP camera. Default is UDP.

Heartbeat Detection If enabled, the PBX will regularly send RTSP OPTIONS to check of the device is still online.

Extensions, Extension Groups, and Departments can be selected to access this IP camera by dialling the configured Device
Allowed Members
Number.
 IPC Devices Settings

RemoteConnect
An integrated & important part of Grandstream’s GDMS cloud-based device management service which runs on Amazon Web Services (AWS) with 99.999%

reliability, the UCM RemoteConnect cloud service supports hassle-free Work-From-Home audio/video communications & collaborations using WebRTC-

based license-free “Grandstream Wave” softphones for desktop/Web/mobile devices (plus GUV series of USB headsets/Webcams), zero-touch out-of-box

automated NAT firewall traversal for remote users & devices, IT-friendly remote management of UCM and attached endpoint devices, and more.

The RemoteConnect can be configured under Other Features🡪RemoteConnect After purchasing the RemoteConnect package.

RemoteConnect

On the GDMS platform, sign in and go to Device🡪PBX Device page, click on “Add Device” to add your UCM6300 device to the GDMS system, once done
an open beta plan will be assigned to the UCM.

In daily operation, the user can click the “Diagnosis” button to diagnose the remote service system. The specific diagnosis content includes media service
(STUN/TURN), GDMS link and heartbeat detection, tunnel service (SIP/Web Socket), Cloud IM, UCM bandwidth speed measurement.
 Remote Diagnosis

UCM RemoteConnect Plan in Effect

After the UCM is added on GDMS, automated NAT traversal, SIP extension sync-up and basic statistics features are available without manual configuration

required.

Plan Settings
After UCM is added into GDMS, all SIP extensions on the UCM will be synced up to GDMS automatically for users to allocate and manage SIP extension for

their end devices. Also, the media NAT Traversal service, alert event sync configuration items are checked by default, the CDR data cloud storage in GDMS
should be manually checked according to user needs.
The UCM supports to allow authorized GDMS user to access UCM without entering the password once the super admin or admin checks “Enable

Passwordless Remote Access”. If super admin enables this option, then the UCM will be accessed using the super admin account. If admin enables this
option, then the UCM will be accessed using the admin account. Super admin and admin can see whether this option is enabled. Additionally, super admin can

disable all accounts who enabled this option while admin can only disable access for the account itself.

The settings are under UCM web GUI: RemoteConnect🡪Plan Settings.

UCM RemoteConnect Plan Settings

After adding UCM to the GDMS platform, UCM will synchronize all SIP extensions to the GDMS platform, this allows to use the GDMS platform for

account allocation and terminal management.

The accounts synchronized to GDMS platform can be viewed on the GDMS-> VoIP Account->SIP Account page. As shown in the figure below:

UCM SIP Extensions synchronized to GDMS

The Media NAT Traversal provides a fully automatic intelligent external network penetration service to ensure that you can make normal calls/conferences on

the external network.

CDR data cloud storage provides a service of dumping to GDMS to prevent CDR from continuously increasing occupying UCM storage space.

Alarm event synchronization is to synchronize the alarm information generated on UCM to the GDMS server.

UCM supports GDMS remote passwordless access. When this button is checked, GDMS remote access UCM does not need to enter the account password,

and no login is required.

Allow the administrator/super administrator to open it. After clicking on open with an account, the subsequent password-free login will use the account. All
administrators and super administrators can see whether this UCM is enabled.

Super administrators can check and uncheck all the exemption lists; administrators can check and uncheck the exemption status of this account, and the

corresponding account exemption access function will be closed after cancellation.

Notes:

Deleting an account on GDMS only removes the association between the account and the device, and does not delete the SIP account information on
UCM.

Any creation, deletion or modification of the SIP account on UCM will be automatically synchronized to the GDMS cloud platform.

After checking the “Media NAT traversal service”, the TURN service and other related traversal settings set by the user will not take effect.

Integrated Customer Service

To configure the Integrated Customer Service, go to the Web GUI 🡪 RemoteConnect 🡪 Integrated Customer Service page that allows users to download

the SDK provided by the customer service system and integrate it on the website, so that the website can contact customer service for call operations. The call
queue is used as the customer service number.

Integrated Customer Service interface

Enabling Click2Call will allow users to initiate a direct call from the web browser by clicking on the call button embedded on the website graphical interface.

The calls initiated can be directed to call queues or a specific extension.

Enterprise UI Customization
With a RemoteConnect plan, on the Web GUI 🡪 RemoteConnect 🡪 Enterprise UI Customization page, users can edit the company name and select a

local image file as the new logo. The company name acts on the text part with the logo, and the pictures are in different formats and sizes according to the
logo position, which are 64*64px (only ico format is supported), 256*256px, 80*80px, which supports users in the “UCM management platform/login” “”,
“Reset Password”, “Email Template”, “Wave_PC”, “Wave Login”, “Browser Label”, “Guide Page” interface preview.
 LOGO 1: Replaces Browser tab icon

LOGO 2: Replaces the Grandstream banner on the top left corner of the management login page and emails.

LOGO 3: Replaces the Grandstream logo on the top left corner of the Wave Web interface and UCM management interface.

Enterprise UI Customization

Statistics
After using UCM RemoteConnect, all remote calls will be logged and concurrent remote calls will be displayed on the UCM. The concurrent remote calls can

be viewed under UCM web GUI🡪Other Features🡪UCM RemoteConnect🡪Statistics page.

Concurrent Remote Calls

For more information, please visit http://ucmrc.gdms.cloud/intro.html and read our UCM63XX RemoteConnect guides
GDMS Cloud Storage Space
When the correspondent RemoteConnect plan is active, the user can access to a GDMS Cloud Storage Space to get an overview about how the storage space

is being used. The data is represented in four categories of file types: CDR Data, Backup Data, Recording Files, and IM Files.

The user is also able to see the names of all the files stored in the GDMS Cloud Storage Space.

GDMS Cloud Storage Space

UCM6300’s Wave App

Wave is a mobile, desktop, and web App that provides remote voice and video collaboration tools for businesses utilizing Grandstream’s UCM6300 series IP
PBXs. For more information, please visit: https://ucmrc.gdms.cloud/intro.html and read our UCM63XX RemoteConnect guides

Grandstream Wave Interface

For more details about the Wave Web, please refer to the following guides:

https://documentation.grandstream.com/knowledge-base/ucm630x-series-wave-web-administration-guide/

https://documentation.grandstream.com/knowledge-base/wave-web-user-guide/

Grandstream Wave is also available in Android/IOS version where it can be downloaded from Google Play or AppStore. For more details, please refer to the

following guide:

https://documentation.grandstream.com/knowledge-base/wave-mobile-application-android-ios-user-guide/

Wave Desktop version is also supported in Windows and MacOS and can be downloaded from: https://fw.gdms.cloud/wave/download/

Download Wave Client

API CONFIGURATION
The UCM630X supports third party billing interface API for external billing software to access CDR and call recordings on the PBX. The API uses HTTPS to
request the CDR data and call recording data matching given parameters as configured on the third-party application.

API Configuration Parameters

Before accessing the API, the administrators need enable API and configure the access/authentication information on the UCM630X first under Other

Features🡪API Configuration. The API configuration parameters are listed in the tables below.

Note

You can create multiple users with different login credentials to access to the HTTPS API.

Note
 The old version of the API interface only supports cdrapi, recapi and pmsapi functions, and will be removed, please use the new HTTPS API instead.

Username The username for API Authentication.

Password The password for API Authentication.

If enabled, 3rd party applications will be able to manage inbound calls via API actions. acceptCall will accept incoming calls
Call Control
while refuseCall will reject them. If no actions are done within 10 seconds, calls will automatically be accepted.

Sets an IP address Access Control List (ACL) for addresses that are allowed to authenticate and register as this user. By default, this is
Permitted IP (s)
not configured, allowing all IP addresses to register to this extension. The format is: "xxx.xxx.xxx.xxx/255.255.255.255

HTTPS API

Settings

(Old)

Basic

Settings

Enable Enable/Disable API. The default setting is disabled.

Configure the IP address for TLS server to bind to. “0.0.0.0” means binding to all interfaces. The port number is optional, and the default

port number is 8443. The IP address must match the common name (host name) in the certificate so that the TLS socket will not bind to
TLS Bind
multiple IP addresses.
Address

The default setting is 0.0.0.0:8443.

Username Configure the username for TLS authentication.

Password Configure the password for TLS authentication.

Specify a list of IP addresses permitted to use the API. This creates an API-specific access control list. Multiple entries are allowed.

Permitted
For example, “192.168.40.3/255.255.255.255” denies access from all IP addresses except 192.168.40.3.
IP(s)

By default, this is blank, which indicates that no IP addresses are allowed to use this API.

Other

Settings

TLS Private
Upload TLS private key. The size of the key file must be under 2MB. This file will be renamed as ‘private.pem’ automatically.
Key

Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate file (*.pem format only) for TLS connection. This
TLS Cert
file will be renamed as “certificate.pem” automatically. It contains private key for the client and signed certificate for the server.
 API Module

CDR API Enable/disable CDR API module.

REC API Enable/disable REC API module.

PMS API Enable/disable PMS API module.

Table 123: Configuration Parameters (Old)

For more details on CDR API (Access to Call Detail Records), REC API (Access to Call Recording Files) and PMS API, please refer the document in the link

here:

https://documentation.grandstream.com/knowledge-base/cdr-rec-api/

https://documentation.grandstream.com/knowledge-base/cdr-rec-api/

PMS API

API Queries Supported

The new API supports now new queries listed below which will accomplish certain requests and get data about different modules on UCM630X.

Queries Supported

getSystemStatus

getSystemGeneralStatus

listAccount

getSIPAccount

updateSIPAccount

listVoIPTrunk

addSIPTrunk

getSIPTrunk

updateSIPTrunk

deleteSIPTrunk

listOutboundRoute

addOutboundRoute

getOutboundRoute

updateOutboundRoute

deleteOutboundRoute

listInboundRoute

addInboundRoute

getInboundRoute

updateInboundRoute
deleteInboundRoute

playPromptByOrg

listBridgedChannels

listUnBridgedChannels

Hangup

callbarge

listQueue

getQueue

updateQueue

addQueue

deleteQueue

loginLogoffQueueAgent

pauseUnpauseQueueAgent

listPaginggroup

addPaginggroup

getPaginggroup

updatePaginggroup

deletePaginggroup

MulticastPaging

MulticastPagingHangup

listIVR

addIVR

getIVR

updateIVR

deleteIVR

cdrapi

recapi

pmsapi

queueapiget

getPinSets

addPinSets

updatePinSets

deletePinSets

cleanTerminalChatInformation

getSIPAccountQR

getCallQueuesMemberMessage
 getQueueCalling

CDR Real-time Output

Settings

Enables real-time CDR output module. This module connects to selected IP addresses and ports and posts CDR strings
Enable
as soon as it is available.

Server Address CDR server IP address

Port CDR server IP port

Upload Prompts User

Configuration

Username Username used to upload prompts.

Password Password used to upload prompts.

Table 125: API Configuration Parameters

Upload Voice Prompt via API

Customers now can use the “Upload Prompts User Configuration” to upload/replace voice prompt files as an alternative method to the manual upload method

on UCM PBX Settings🡪 Voice Prompt🡪 Custom Prompt.

The workflow of the prompt file upload goes as:

An HTTP/HTTPS request is sent to the UCM to upload/replace a voice prompt file, the request should include authentication details to the UCM and the

name of the file to be uploaded. Then the UCM will contact an FTP server that should be hosted on the same IP address of the HTTP/HTTPS requester and

download the prompt file from the FTP server.

The steps and conditions to upload the voice prompt via API are listed below:

1. Configure the prompt User under Other Features 🡪 API Configuration 🡪 Upload Prompts User Configuration. By default, the username and
password for voice prompt user are “Username: uploader; Password: uploader123”.

Upload Prompt User Configuration

2. Hash the password of the user configured to an MD5 Encryption format.

3. Set the permission on the FTP server to Anonymous on the local computer hosting the FTP server and make sure that the default FTP port 21 is used.
 4. Send an HTTP/HTTPS command to trigger the Prompt file upload on the UCM. If UCM’s HTTP server is set to HTTPS, the example of the request sent

to the UCM is:

https://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c21b3aabd779213c7179462&filename=test.mp3

If UCM’s HTTP server is set to HTTP, the example of the request sent to the UCM is：

https://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c21b3aabd779213c7179462&filename=test.mp3

Note: If the File name on the HTTP/HTTPS request exists already on the UCM’s Custom voice prompts list the existing file will be overwritten by the new

file downloaded from the FTP server.

For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording Files), please refer the document in the link here:

https://documentation.grandstream.com/knowledge-base/cdr-rec-api/

CTI SERVER
UCM does support CTI server capabilities which are designed to be a part of the CTI solution suite provided by Grandstream, including GXP21XX and

GXP17XX enterprise IP phones along with GS Affinity app.

Mainly the UCM will by default listening on port TCP 8888 for the connections from GS affinity application in order to interact, modify and serve data

requests by the application which includes setting call features for the connected extension as call forward and DND.

Users can change the listening port under the menu page, Web GUI🡪Other Features🡪CTI Server as shown on below screenshot:

CTI Server Listening port

More information about GS affinity and CTI Support on Grandstream products series please refer to the following link:

https://documentation.grandstream.com/knowledge-base/gs-affinity-user-guide/

ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)

The UCM630X supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands

or read events over a TCP/IP stream. It is particularly useful when the system admin tries to track the state of a telephony client inside Asterisk.

User could configure AMI parameters on UCM630X Web GUI🡪Other Features🡪AMI. For details on how to use AMI on UCM630X, please refer to the
following AMI guide:

https://documentation.grandstream.com/knowledge-base/ami-asterisk-management-interface/

Please do not enable AMI on the UCM630X if it is placed on a public or untrusted network unless you have taken steps to protect the device from unauthorized

access. It is crucial to understand that AMI access can allow AMI user to originate calls and the data exchanged via AMI is often very sensitive and private for

your UCM630X system. Please be cautious when enabling AMI access on the UCM630X and restrict the permission granted to the AMI user. By using AMI on

UCM630X you agree you understand and acknowledge the risks associated with this.

CRM INTEGRATION
Customer relationship management (CRM) is a term that refers to practices, strategies and technologies that companies use to manage and analyze

customer interactions and data throughout the customer lifecycle, with the goal of improving business relationships with customers.

The UCM630X support the following CRMs: SugarCRM, VtigerCRM, ZohoCRM, Salesforce CRM and ACT! CRM, which allows users to look for contact

information in the Contacts, Leads and / or Accounts tables, shows the contact record in CRM page, and saves the call information in the contact’s history.

Sugar CRM
Configuration page of the SugarCRM can be accessed via admin login, on the UCM WebGUI🡪Other Features🡪CRM.

SugarCRM Basic Settings

1. Select “SugarCRM” from the CRM System Dropdown in order to use SugarCRM.

Select a CRM system from the dropdown menu, four CRM systems are available: Sugar CRM, Vtiger CRM, ZohoCRM (v1),
CRM System
Salesforce and ACT! CRM.

CRM Server
Enter the IP address of the CRM server.
Address

Add Unknown
Add the new number to this module if it cannot be found in the selected module.
Number

Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables,
Contact Lookups
Leads, Accounts, and Contacts.

Table 126: SugarCRM Settings

Once settings on admin access are configured:

2. Click on and .

3. Logout from admin access.

4. Login to the UCM as user and navigate under “User Portal🡪Other Feature🡪CRM User Settings”.

Click on “Enable CRM” and enter the username/password associated with the CRM account then click on and . The status will

change from “Logged Out” to “Logged In”. User can start then using SugarCRM features.
 CRM User Settings

Vtiger CRM
Configuration page of the VtigerCRM can be accessed via admin login, on the UCM WebGUI🡪Other Features🡪CRM.

VtigerCRM Basic Settings

1. Select “Vtiger CRM” from the CRM System Dropdown in order to use Vtiger CRM.

Select a CRM system from the dropdown menu, four CRM systems are available: Sugar CRM, Vtiger CRM, Zoho CRM (v2),
CRM System
Salesforce and ACT! CRM.

CRM Server
Enter the IP address of the CRM server.
Address

Add Unknown
Add the new number to this module if it cannot be found in the selected module.
Number

Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables,
Contact Lookups
Leads, Organizations, and Contacts.

Table 127: VtigerCRM Settings

Once settings on admin access are configured:

 2. Click on and .

3. Logout from admin access.

4. Login to the UCM as user and navigate under “User Portal🡪Other Features🡪CRM User Settings”.

Click on “Enable CRM” and enter the username/password associated with the CRM account then click on and . The status will
change from “Logged Out” to “Logged In”. User can start then using SugarCRM features.

CRM User Settings

Zoho Telephony
Due to changes related to authenticating with ZohoCRM, Zoho’s CRM integration setup process has been updated. The Other Features→CRM page will

appear as such when Zoho Telephony is selected.

Configure Zoho Telephony

Once the desired settings have been configured, save and apply changes. Next, log into the User Portal and navigate to Other Features→CRM User Settings.
 CRM User Settings

Click on the CRM Validation button, and the user will be redirected to the following page:

Enter your CRM account details and log in.

Click on Accept.

Return to the UCM User Portal page. The CRM User Settings page should now look like the following:

CRM User Login Status

Users can then click on the Enable Integrate and Enable Click-to-Dial buttons to fully enable Zoho Telephony integration, which allows users to click on a

call button next to contacts in the CRM Contacts page to initiate calls between UCM extensions and CRM contacts.
 Call Client

Installing Zoho Phonebridge

In the scenario where an error is received saying that Phonebridge has not been enabled, users may need to reinstall it.

Error Message

To do so, go to https://crm.zoho.com/ click on the Settings icon in the top right corner of the Zoho CRM page.

Zoho Settings

Under the Channels section, click on Telephony.

Zoho Telephony Channels

The following page will appear:

 Zoho Telephony MarketPlace

Click on the search bar in the top right and search for “Grandstream”. The following entry should appear:

Telephony MarketPlace

Click on it to see the following:

Grandstream Networks Integration Installation

Finally, click on the green Install button.

 Integration Installed

Zoho Telephony Phonebridge is now installed. Users can now go back to the UCM User Portal to enable the integration and click-to-dial.

Installing Zoho Wave Add-in as an Alternative

Wave Desktop supports the installation of an add-in that works with UCM’s Zoho Telephony integration to provide convenient access to CRM contact
information. This is redundant with setting up integration from the User Portal. If Wave is used, it would be best to set up Zoho integration using only this

method instead of the steps mentioned above. The add-in can be installed from the Wave App Store.

Wave Desktop Zoho CRM Add-in

After selecting the desired server and clicking the Identity Authorization button, the following page will appear:
 Zoho Add-in Permissions

Click on Accept. The Wave authentication window should now change to this:

Log into Zoho CRM

Click on the Log In button. The following will be displayed:

 Zoho CRM Information

From here, users will be able to select the information they would like to see on call notification cards when receiving calls from CRM contacts. This

information can be modified from the CRM portal.

Add Contact

If the caller is an unknown number, users can add it as a new contact into the CRM system. Clicking on the Add Contact button will redirect users to the

Zoho CRM portal’s Create Contact page, where users can manually fill in the contact information.

Create Contact
 Note

It seems like Wave cannot automatically fill in the phone number information of an unknown caller. However, if the UCM admin portal’s CRM settings were

configured to automatically add unknown numbers as contacts, then this step would be redundant as the contact would already be created upon receiving the call.

The users would need to locate the newly created contact in the Contacts page and edit the contact information accordingly.

Salesforce CRM
Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM Web GUI🡪Other Features🡪CRM”.

Salesforce Basic Settings

1. Select “Salesforce” from the CRM System Dropdown in order to use Salesforce CRM.

Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, VtigerCRM, ZohoCRM (v1&v2),
CRM System
Salesforce and ACT! CRM.

Add Unknown
Add the new number to this module if it cannot be found in the selected module.
Number

Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables,
Contact Lookups
Leads, Accounts, and Contacts.

Table 129: Salesforce Settings

Once settings on admin access are configured:

2. Click on and .

3. Logout from admin access.

4. Login to the UCM as user and navigate under “User Portal🡪Other Features🡪CRM User Settings”.

Click on “Enable CRM” and enter the username, password and Security Token associated with the CRM account then click on and

. The status will change from “Logged Out” to “Logged In”. User can start then using Salesforce CRM features.
 Salesforce User Settings

ACT! CRM
Configuration page of the ACT! CRM can be accessed via admin login, on the UCM Web GUI🡪Other Features🡪CRM”.

The configuration steps of the ACT! CRM are as follows:

1. Navigate to Other Features🡪CRM and select the “ACT! CRM” option.

Enabling ACT! CRM

2. Log into the UCM as a regular user and navigate to Other Features🡪CRM User Settings and check “Enable CRM” option and enter the username and

password, which will be the ACT! CRM account’s API Key and Developer Key, respectively. To obtain these, please refer to the ACT! CRM API
developer’s guide here: https://mycloud.act.com/act/Help

Enabling CRM on the User Portal

PMS INTEGRATION
UCM630X supports Hotel Property Management System PMS, including check-in/check-out services, wakeup calls, room status, Do Not Disturb which

provide an ease of management for hotel applications. This feature can be found on Web GUI🡪Other Features🡪PMS.
 The PMS integration on UCM is currently supported only with one of the four following solutions.

The PMS module built-in the UCM supports the following features based on each solution:

Feature Mitel HMobile HSC IDS

Check-In ✓ ✓ ✗ ✓

Check-out ✓ ✓ ✗ ✓

Wake-up Call ✓ ✓ ✗ ✓

Name Change ✓ ✗ ✓ ✗

Update ✗ ✓ ✗ ✓

Set Credit ✓ ✗ ✗ ✗

Set Station Restriction ✓ ✗ ✓ ✗

Room Status ✗ ✓ ✗ ✓

Room Move ✗ ✓ ✗ ✓

Do Not Disturb ✗ ✓ ✓ ✗

Mini Bar ✗ ✓ ✗ ✓

MSG ✗ ✓ ✗ ✗

MWI ✗ ✗ ✓ ✗

Unconditional Call Forward ✗ ✗ ✓ ✗

Table 130: PMS Supported Features

HMobile PMS Connector

In this mode, the system can be divided into three parts:

PMS (Property Management System)

PMSI (Property Management System Interface)

PBX

Grandstream UCM6XXX series have integrated HMobile Connect PMSI which supports a large variety of PMS software providing following hospitality
features: Check-in, Check-out, set Room Status, Wake-up call and more.
The following figure illustrates the communication flow between the UCM and PMS software, which is done through a middleware system (HMobile

Connect) acting as interface between both parties.

UCM & PMS interaction

HSC PMS
In this mode, the system can be divided into two parts:

PMS (Property Management System)

PBX

Grandstream UCM6XXX series have integrated HSC PMS providing following features:

Changing Display Name

Set Station Restriction

Call forwarding

DND

Name Change

MWI

Note:

1. Added support for receiving HTTP GET keep-alive messages from HSC PMS. This will allow the PMS to be aware of its connection to the UCM and

take the appropriate actions such as raising alarms, sending notifications, etc.

2. Added support for HTTP GET requests from HSC PMS to retrieve UCM extension information. UCM can provide the following information:

extension – UCM extension number

name – extension display name / CID name

mwi – MWI state

permission – permission level of the extension

cfwt – call forwarding always number

 dnd – DND state

language – display language of the extension in ISO 639-1 format

The UCM should respond with either 200 OK or 404 responses.

1. Added HTTPS support

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software (HSC). The communication

between both parties is direct with no middleware.

UCM & HSC PMS interaction

Mitel PMS
In this mode, the system can be divided into two parts:

PMS (Property Management System)

PBX

Grandstream UCM6XXX series have integrated Mitel PMS providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and

more.

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software (Mitel). The communication
between both parties is direct with no middleware.

Figure 276: UCM & Mitel PMS interaction

IDS PMS
In this mode, the system can be divided into two parts:

PMS (Property Management System)

PBX

The Grandstream UCM series integrates IDS PMS to set room status, Mini Bar, wake up calls, activate/deactivate dialing permissions, and more.

Figure 277: UCM & IDS PMS interaction

PMS API
The PMS API allows users to use their own middleware to work with PMS systems instead of currently supported integrations.

Additionally, this API allows access to read and modify certain UCM parameters that current supported PMS integrations cannot. To use this, users must first

enable and configure the HTTPS API settings.

For more details, please refer to online https://documentation.grandstream.com/knowledge-base/https-api/, Pmsapi section.

Connecting to PMS
On the UCM WebGUI🡪Other Features🡪PMS🡪Basic Settings” set the connection information for the PMS platform.

Field Description

Users can select the desired PMS module from the drop-down list.

● Hmobile

● Mitel
PMS Module
● HSC

● IDS

● PMS API

A customized prompts that can be played when the wakeup call is answered. To customize it please navigate to PBX Settings 🡪
Wakeup Prompt
Voice Prompt 🡪 Custom Prompt
 Username This username is used to authenticate into the PMS API.

Password This password is used to autheticate into the PMS API.

Back Up Voicemail
Back up voicemail recordings to external storage after check-out.
Recordings

Scheduled wakeup calls for rooms can be cleared upong checking in or checking out.

Automatically Clear ● None: The wakeup calls won't be automatically cleared.

Wakeup Calls ● Check out: The wake up calls assigned to the guest will be cleared when they check out.

● Check In: The wake up calls assigned to a guest will be cleared when a new client checks in.

Automatically Clear Wave

If enabled, room Wave chat history will be automatically cleared upon check-in or check-out.
Chat History

Automatically Reset
If enabled, the User/Wave password of the room extension will be automatically reset to a random password upon check-out.
User/Wave Password

In order to use some PMS features please activate the feature code associated under “Call Features🡪Feature Codes”

Update PMS Room Status

PMS Wake Up Service

PMS Features

Room Management

In Room Management tab, the user can create a room and affect up to two extensions to it. This will appear in Room Status tab, and from there the user can
change the Check-in/Check-out

Create New Room

Call Privileges allows the administrator to set the level of call privilege of the room.
Room Status

Figure 278: Check-in a Client

After clicking “OK” the client entry will be added to the list.

The user can click on Check-in/Check-out Records to view the history of the checked-in and checked-out guests.

Note

The Call Privilege configured during a guest’s check-in will be reset to the room’s default call privilege upon guest check-out.

Wake Up Service

In order to create a New Wake up service, user can click on ”Add”, the following window will pop up:

Create New Wake Up Service

 Field Description

Room Number Select the room number where to call with a limitation of 63 characters.

Select Time Set the time of the wakeup call

Show the status of the call:

Programmed: the call is scheduled for the time set

Action Status Cancelled: the call is canceled

Executed: the wakeup call is made

Note: Editing an already executed wakeup service will automatically change the service’s status to “Programmed”.

Single: The call will be made once on the specific time.

Type
Daily: The call will be repeated every day on the specific time

Table 132: PMS Wake up Service

Once the call is made on the time specified, the following figure show the status of the wakeup call.

Wakeup Call executed

This call has been executed but has been rejected, that why we can see the “Busy” status.

Maid

In order to create a new maid, click on under UCM WebGUI🡪Other Features🡪PMS🡪Maid.

Create New Maid

Maid Code Enter the Code to use when the maid wants to use the Mini Bar.

Password Enter the password associated with the maid.

Table 134: Create New Maid

In order to create a new consumer goods, click on under UCM WebGUI🡪Other Features🡪PMS🡪Mini Bar, the following
window will popup.
 Create New Consumer Goods

Code Enter the Goods Code.

Name Enter the Name of the Goods

The Minibar page displays as:

Mini Bar

Note

Please note that you can dial the “Update PMS Room Status” feature code, the “Maid” feature code, and the “Room Status” feature code all at once to change the

room status.

Local PMS
UCM6300 series offer a local Property Management System to give the user basic management features without having to purchase a PMS for the most basic

property management actions. In addition to Room Management, Rooms Status for checking-in and checking-out, Wakeup Service, Mini Bar, and Maid
functions, the UCM6300 allows a number of additional functions upon checking-out, like backing up voicemail recordings, clearing wakeup calls and Wave

history automatically, in addition to resetting Wave’s password. The user can use the Local PMS feature to check-in and check-out clients from the web user

interface.
 Local PMS

Field Description

Users can select the desired PMS module from the drop-down list.

● Hmobile

● Mitel
PMS Module
● HSC

● IDS

● PMS API

A customized prompts that can be played when the wakeup call is answered. To customize it please navigate to PBX Settings 🡪
Wakeup Prompt
Voice Prompt 🡪 Custom Prompt

Username This username is used to authenticate into the PMS API.

Password This password is used to autheticate into the PMS API.

Back Up Voicemail
Back up voicemail recordings to external storage after check-out.
Recordings

Scheduled wakeup calls for rooms can be cleared upong checking in or checking out.

Automatically Clear ● None: The wakeup calls won't be automatically cleared.

Wakeup Calls ● Check out: The wake up calls assigned to the guest will be cleared when they check out.

● Check In: The wake up calls assigned to a guest will be cleared when a new client checks in.

Automatically Clear Wave

If enabled, room Wave chat history will be automatically cleared upon check-in or check-out.
Chat History

Automatically Reset
If enabled, the User/Wave password of the room extension will be automatically reset to a random password upon check-out.
User/Wave Password
WAKEUP SERVICE
The Wake Up service can be used to schedule a reminder or wake up calls to any valid destination. This service is available on the UCM630X as a separated

module.

There are three ways to set up Wakeup Service:

Using admin login

Using user portal

Using feature code

Wake Up Service using Admin Login

1. Login to the UCM as admin.

2. Wake Up service can be found under Web GUI🡪Other Features🡪Wakeup Service, click on ”Add” to create a new wakeup service. The following
window will pop up.

Create New Wakeup Service

3. Fill out the required fields and select the members to add to the wakeup group.

Enable Wakeup Service Enable Wakeup service.

Name Enter a name (up to 64 characters) to identify the wakeup service.

Prompt Select the prompt to play for that extension.

 If disabled, users can select a specific date and time.
Custom Date
If enabled users can select multiple days of the week to perform the wakeup.

Date Select the date or dates when to performs the wakeup call.

Time Select the time when to play the wakeup call.

Members Select the members involved within the wakeup service group.

Table 135: Wakeup Service

4. Click and to apply the changes.

A wakeup service entry is created. The UCM will send a wakeup call to every extension in the member list at the scheduled date and time.

Note: the wakeup service has the following limitation on how many members can be added depending on UCM model.

UCM Model Max members in a Wakeup Service

UCM6301 50

UCM6302 100

UCM6304 150

UCM6308 200

Table 136: Max Wakeup Members

Wake Up Service from User Portal

1. Login to the user portal on the UCM630X.

2. Wake Up service can be found under “Other Features🡪Wakeup Service”, click on ”Add” to create a new wakeup service.

3. Configures the Name, Prompt, Date and Time for the user to make the wakeup to.

4. Click and to apply the changes.

Wake Up Service using Feature Code

1. Login to the UCM as admin.

2. Enable “Wakeup Service” from the Web GUI under “Call Features🡪Feature Codes”.
 Wakeup Service Feature Code

3. Click and to apply the changes.

4. Dial “*36” which is the feature code by default to access to the UCM wakeup service to add, update, activate or deactivate UCM wakeup service.

ANNOUNCEMENT CENTER
The UCM630X supports Announcement Center feature which allows users to pre-record and store voice message into UCM630X with a specified code. The

users can also create group with specified extensions. When the code and the group number are dialed together in the combination of code + group number,

the specified voice message is sent to all group members and only extensions in the group will hear the voice message.

Figure 288: Announcement Center

Announcement Center Settings

Name Configure a name for the newly created Announcement Center to identify this announcement center.

Enter a code number for the custom prompt. This code will be used in combination with the group number. For example, if

the code is 55, and group number is 666. The user can dial 55666 to send prompt 55 to all members in group 666.
Code
Note: The combination number must not conflict with any number in the system such as extension number or conference

number.

This option is to set a custom prompt as an announcement to notify group members. The file can be uploaded from page
Custom Prompt
‘Custom Prompt’. Click ‘Prompt’ to add additional record.

Ring Timeout Configure the ring timeout for the group members. The default value is 30 seconds.

Auto Answer If set to Yes, the Auto answer will be enabled by the members.
Group Settings

Configure a name for the newly created group to identify the group.
Name
Note: Name cannot exceed 64 characters.

Configure the group number. The group number is used in combination with the code. For example, if group number is 666, and

code is 55. The user can dial 55666 to send prompt 55 to all members in group 666.
Number
Note: The combination number must not conflict with any number in the system such as extension number or conference

number and cannot exceed 64 characters.

Members Select the group members from the available list.

Announcement Center feature can be found under Web GUI🡪Other Features🡪Announcement Center. The following example demonstrates the usage of
this feature.

1. Click to add new group.

2. Give a name to the newly created group.

3. Create a group number which is used with code to send voice message.

4. Select the extensions to be included in the group, who will receive the voice message.

Figure 289: Announcement Center Group Configuration

In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.

5. Click to create a new Announcement Center.

6. Give a name to the newly created Announcement Center.

7. Specify the code which will be used with group number to send the voice message to.

8. Select the message that will be used by the code from the Custom Prompt drop down menu. To create a new Prompt, please click “Prompt” link and
follow the instructions in that page.
 Figure 290: Announcement Center Code Configuration

Code and Group number are used together to direct specified message to the target group. All extensions in the group will receive the message. For example,

we can send code 55 to group 666 by dialing 55666 from any extension registered to the UCM630X. All the members in group 666 which are extension 1000,

1001 and 1002 will receive this voice message after they pick up the call.

Figure 291: Announcement Center Example

QUEUEMETRICS
The QueueMetrics docking tool provides an interface for UCM system and QM docking. Pass the UCM call queue report to QueueMetrics in a richer form.

QueueMetrics is a call center control platform that supports login and logout of frequently used agents in the call center, provides call reports, real-time queue

monitoring and other functions.

Figure 292: QueueMetrics

Enable QueueMetrics
Disabled by default.
Integration

Enter the URL of the QueueMetrics on-premise server you have installed. (i.e. http://xxx.xxx.xxx.xxx:8080/queuemetrics.).

QueueMetrics URL
Note: Under normal circumstances, the user is a webqloader type user of Queue Metrics. You must ensure that the user is

enabled, otherwise the connection will fail.

 Username Configure the username

Webqloader Password Configure the user password.

Partition Configure the data storage partition identifier.

Table 139: QueueMetrics configuration parameters

STATUS AND REPORTING

PBX Status
The UCM630X monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces, Storage Device Life (eMMC lifetime), the number of
public/random video meeting and Parking lot. It presents administrators the real-time status in different sections under Web GUI🡪System

Status🡪Dashboard.

Figure 293: Status🡪PBX Status

Trunks

Users could see all the configured trunk status in this section.

Figure 294: Trunk Status

Status Display trunk status.

● Analog trunk status:

 - Available

- Busy

- Unavailable

- Unknown Error

● SIP Peer trunk status:

- Unreachable: The hostname cannot be reached.

- Unmonitored: Heartbeat feature is not turned on to be monitored.

- Reachable: The hostname can be reached.

● SIP Register trunk status:

- Registered

- Unrecognized Trunk

Trunks Display trunk name

Display trunk Type:

● Analog
Type
● SIP

● IAX

Username Display username for this trunk.

Port/Hostname/IP Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.

Extensions

Extensions Status can be seen from the same configuration page, users can go under Web GUI🡪Extension/Trunk🡪Extensions and following page will be
displayed listing the extensions and their status information.

Figure 295: Extension Status

Display extension number (including feature code). The color indicator has the following definitions.

Green: Free

Status Blue: Ringing

Yellow: In Use

Grey: Unavailable

Presence Status Display the presence status of the extension.

Extension Display the extension number.

 Name First name and last name of the extension.

IP and Port Display the IP and port number of the registered device.

Display Email Notification status for the extension.

Email
When notification is waiting for be sent, shows and once sent it will display

Displays extension type.

SIP User

IAX User
Terminal Type
Analog User

Ring Groups

Voicemail Groups

Table 141: Extension Status

Interfaces Status

This section displays interface/port connection status on the UCM630X. The following example shows the interface status for UCM6304 with USB, WAN

port, FXS1, FXS2 and FXO1 connected.

Figure 296: UCM6304 Interfaces Status

USB connected.

USB disconnected.

SD Card connected.

SD Card disconnected.
 LAN/WAN connected.

LAN/WAN not configured.

LAN/WAN disconnected.

FXS/FXO connected.

FXS/FXO waiting.

FXS/FXO busy.

FXS/FXO not configured.

FXS/FXO disconnected.

Table 142: Interface Status Indicators

System Status
The UCM630X system status can be accessed via Web GUI🡪Status🡪System Status, which displays the following system information.

General

Under Web GUI🡪System Status🡪System Information🡪General, users could check the hardware and software information for the UCM630X. Please see
details in the following table.

System Status 🡪System Information🡪General

Model Product model.

Part Number Product part number.

System Time Current system time. The current system time is also available on the upper right of each web page.

Up Time System up time since the last reboot.

Boot Boot version.

Core Core version.

Base Base version.

 Program Program version. This is the main software release version.

Recovery Recovery version.

Lang Lang version

Wave Wave version

Table 143: System Status🡪General

Network

Under Web GUI🡪System Status🡪System Information🡪Network, users could check the network information for the UCM630X. Please see details in the
following table.

System Status🡪System

Status🡪Network

Global unique ID of device, in HEX format. The MAC address can be found on the label coming with original box and
MAC Address
on the label located on the bottom of the device.

IPv4 Address IPv4 address.

IPv6 Address Link IPv6 address

Gateway Default gateway address.

Subnet Mask Subnet mask address.

DNS Server DNS Server address.

Duplex Mode Duplex Mode

Speed Speed

Table 144: System Status🡪Network

Remark

The UCM admin could add remark on UCM web UI->System Status->System Information->Remark to log any necessary information for the UCM such as

location, technical contacts, important topology information and etc. This could be useful for UCM admin especially when there are multiple UCMs to be

managed.

If this UCM has UCMRC service, the remark will also be sync up to GDMS. If this information is edited on GDMS, it will also be updated to the UCM web

UI.
 Figure 297: System Status🡪System Information 🡪 Remark

Storage Usage

Users could access the storage usage information from Web GUI🡪System Status🡪Dashboard🡪Storage Usage. It shows the available and used space for
Space Usage and Inode Usage.

Space Usage includes:

Configuration partition

This partition contains PBX system configuration files and service configuration files.

Data partition

Voicemail, recording files, IVR file, Music on Hold files etc.

USB disk

USB disk will display if connected.

SD Card

SD Card will display if connected.

Inode Usage includes:

Configuration partition

Data partition

Note:

Inode is the pointer used for file reference in the system. The system usually has limited resources of pointers
 Figure 298: System Status🡪Storage Usage

Resource Usage

When configuring and managing the UCM630X, users could access resource usage information to estimate the current usage and allocate the resources

accordingly. Under Web GUI 🡪 System Status 🡪 Dashboard 🡪 Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.

Figure 299: System Status🡪Resource Usage

System Events
The UCM630X can monitor important system events, log the alerts, and send Email notifications to the system administrator.

Alert Events List

The system alert events list can be found under Web GUI 🡪 Maintenance 🡪 System Events. The following event and their actions are currently supported
on the UCM630X which will have alert and/or Email generated if occurred:
 Figure 300: Alert Event List

Alert Events

Fail2ban Blocking

Flood Attacks

Network Traffic Storm

User Login Banned

Remote Login

User Login Success

User Login Failed

System Crash

Restore Config

System Update

System Reboot

TLS Cert Expiration

HA Failure Warning

Cloud IM Abnormal
 Modify Super Admin Password

Data Sync Backup

Local Disk Usage

External Disk Usage

Extended Disk Status

Emergency Calls

SIP Outgoing Call through Trunk Failure

SIP Internal Call Failure

Remote Concurrent Calls

Excessive Outbound Calls

Long Outbound Call Duration

Trunk Outbound Call Duration Usage

Trunk Concurrent Calls

Register SIP trunk failed

SIP Peer Trunk Status

Register SIP failed

SIP Lost Registration

For users who have purchased a GDMS package, once the option Alert Events Sync is enabled under RemoteConnect, the triggered events will be pushed to their
GDMS platform.

Click on to configure the parameters for each event. See examples below.

1. Fail2ban blocking: If the system Fail2ban is blocking, the event will be recorded in the alert log.

2. Flood Attacks: An alert will be generated in case a DDoS attack attempt is detected by the UCM. The event will be registered in the alert log and it will

be pushed to the GDMS.

3. Network Traffic Storm: An alert will be generated in case there is an excessive amount of packets on the LAN. Network Traffic Storms consume the

resources of the network components and saturate the bandwidth, which will bring the whole network to a halt. This event will be registered in events log

and a notification will be pushed to the GDMS.

4. User login banned: If user login is blocked, the event will be recorded in the alert log.
 5. Remote Login: An alert will be generated upon a remote login.

6. System Crash:

Figure 301: System Events🡪Alert Events Lists: System Crash

Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can enter the number and then select

second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Click on the switch to turn on/off the alert and Email notification for the event. Users could also select the checkbox for each event and then

click on button “Alert On”, “Alert Off”, “Email Notification On”, “Email Notification Off” to control the alert and Email notification configuration.

7. Restore Config: Once the system configuration is restored, the configuration restoration event will be recorded in the alert log.

8. System Update: Once the system is upgraded, the system upgrade event will be recorded in the alarm log.

9. System Reboot: UCM will detect the system restart and will send an alert for it. There are two kinds of reboots that the UCM detects, normal and abnormal

reboots. Normal reboots are the reboots that are done when you press the restart button on the web UI, reboot that occur after updating the firmware, HA
backup reboot etc… Abnormal reboots are the reboots that occur due to a system failure. Normal reboots are registered in the alert log and they are not pushed

to GDMS, while abnormal reboots are registered in the alert list and are pushed to GDMS.

10. TLS Cert Expiration: Starting 7 days before the HTTP Server TLS certificate in the UCM device expires, an expiration countdown notification is sent

every day; the certificate has expired, an expiration notification is sent; after the alarm notification is generated, a valid new certificate is uploaded, and a

notification to restore the TLS certificate is generated.

11. HA failure warning: After the HA dual-system hot backup disaster recovery function is enabled in the UCM device, the HA fault alarm is automatically

turned on. When the device has a software and hardware related fault, an HA fault alarm is generated.

12. Cloud IM Abnormal: An alert message will be generated f the Cloud IM encounter any issue or exhibit any abnormal behavior.

13. Modify Super Admin Password: Once the super administrator password is modified, the system will record the password modification event in the alarm

log.

14. NAS: If the system network disk is abnormal, the event will be recorded in the alarm log.

15. Disk Usage

Figure 302: System Events🡪Alert Events Lists: Disk Usage

Detect Cycle: The UCM630X will perform the internal disk usage detection based on this cycle. Users can enter the number and then select
second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
 Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.

If the threshold is exceeded, any behavior of operating the disk will be rejected, including stopping file upload, IM writing, recording and CDR recording.

16. Memory Usage

Figure 303: System Events🡪Alert Events Lists: Memory Usage

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.

17. External Disk Usage

Figure 304: System Events🡪Alert Events Lists: External Disk Usage

Detect Cycle: The UCM630X will perform the External disk usage detection based on this cycle. Users can enter the number and then select

second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.

18. External disk status: If the external disk of the system is Connected/Disconnected, the event will be recorded in the alarm log.

19. CPU Usage Call Control: The CPU flow control threshold is defined under System Settings 🡪 General Settings, and the default value is 90%. When
the traffic exceeds the predetermined value, the event will be recorded in the alert log and new calls will be prohibited.

20. Emergency Calls: If the system generates an emergency call, the event will be recorded in the alert log.

21. Register SIP trunk failed

Figure 305: System Events🡪Alert Events Lists: Register SIP Trunk Failed

Detect Cycle: The UCM will detect the failure of SIP trunk registration at a set interval. Users can enter the number and then select
second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

22. SIP peer trunk status: If the SIP peer trunks status is abnormal, the event will be recorded in the alert log.

23. SIP Outgoing Call through Trunk Failure: If the system SIP trunk outgoing call fails, the event will be recorded in the alert log.
24. Register SIP failed

Figure 306: System Events🡪Alert Events Lists: Register SIP Failed

Configure the sending period of the SIP registration failure alert. The first registration failure alert of the same IP to the same SIP account will be sent

immediately, and then no alerts will be sent for similar failure warnings in the cycle time. After the cycle time expires, an alert will be sent again to count the

number of occurrences of similar SIP registration failure alerts during the cycle. When set to 0, alerts are always sent immediately.

25. SIP lost registration: If System SIP extension registration is lost, the event will be recorded in the alert log.

26. SIP Internal Call Failure: If the system SIP extension call fails within the office, the event will be recorded in the alert log.

27. High Frequency Outgoing Call: When an extension initiates calls frequently, an alert will be logged in the alert log and a notification will be pushed to

the GDMS and through email as well.

28. Remote concurrent calls: If the remote concurrent call fails, the event will be recorded in the alert log.

29. Trunk Outbound Call Duration Usage:

30. Trunk Concurrent calls: When the system detects that the number of concurrent calls of a certain relay exceeds the threshold set by the relay within a

certain period of time, the event will be recorded in the alarm log. Calls are not restricted if the threshold is exceeded.

31. User login success: Successful user login events will be recorded in the alert log.

32. User login failed: User login failure events will be recorded in the alert log.

33. Data Sync Backup: If the system performs data synchronization and backup abnormalities, the event will be recorded in the alert log.

Alert Log

Under Web GUI🡪Maintenance🡪System Events🡪Alert Log, system messages from triggered system events are listed as alert logs. The following
screenshot shows system crash alert logs.

Figure 307: System Events🡪Alert Log

User could also filter alert logs by selecting a certain event category, type of alert log, and/or specifying a certain time period. The matching results will be

displayed after clicking on . Alert logs are classified into two types by the system:

1. Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the alert threshold.

2. Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping back below the alert threshold.

User could filter out alert logs of “Generate Alert” or “Restore to Normal” by specifying the type according to need. The following figure shows an example
of filtering out alert logs of type of “Restore to Normal”.
 Figure 308: Filter for Alert Log

Alert Contact

This feature allows the administrator to be notified when one of the Alert events mentioned above happens. Users could add administrator’s Email address

under Web GUI🡪Maintenance🡪System Events🡪Alert Contact to send the alert notification to an email (Up to 10 Email addresses can be added) or also
specify an HTTP server where to send this alert.

Configure the email addresses to send alert notifications to.

Super Admin

Email
Up to 10 email addresses can be added.

Configure the email addresses to send alert notifications to.

Admin Email
Up to 10 email addresses can be added.

Email Template Please refer to section [Email Templates]

Protocol used to communicate with the server. HTTP or HTTPS.

Protocol
Default one is HTTP.

HTTP Server The IP address or FQDN of the HTTP/HTTPS server.

HTTP Server
HTTP/HTTPS port
Port

Customize the template used for system warnings.

Warning

Template
By default: {“action”:”${ACTION}”,”mac”:”${MAC}”,”content”:”${WARNING_MSG}”}

Customize the notification template to receive relevant alert information.

By default:
Notification
{“action”:”${ACTION}”,”cpu”:”${CPU_USED}”,”memery”:”${MEM_USED}”,”disk”:”${DISK_USED}”,”external_disk”:”${E
Template
XTERNAL_DISK_USED}”}

Note: The notification message with “action:0” will be sent periodically if Notification Interval is set.
 Modifies the frequency at which notifications are sent in seconds.
Notification

Interval
No notifications will be sent if the value is “0”. Default value: 20

${MAC} : MAC Address

${WARNING_MSG} : Warning message

${TIME} : Current System Time

${CPU_USED} : CPU Usage

Template

Variables
${MEM_USED} : Memory Usage

${ACTION} : Message Type. Refer to [Table 145: Alert Events]

${DISK_USED} : Disk Usage

${EXTERNAL_DISK_USED} : Disk Usage

Table 146: Alert Contact

CDR
CDR (Call Detail Record) is a data record generated by the PBX that contains attributes specific to a single instance of phone call handled by the PBX. It has

several data fields to provide detailed description for the call, such as phone number of the calling party, phone number of the receiving party, start time, call

duration, etc.

On the UCM630X, the CDR can be accessed under Web GUI🡪CDR🡪CDR. Users could filter the call report by specifying the date range and criteria,
depending on how the users would like to include the logs to the report. Click on “Filter” button to display the generated report.

Figure 309: CDR Filter

Groups the following:

Inbound calls: Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and sent to an internal extension.

Call Type Outbound calls: Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an internal extension.

Internal calls: Internal calls are calls from one internal extension to another extension, which are not sent over a trunk.

External calls: External calls are calls sent from one trunk to another trunk, which are not sent to any internal extension.
 Filter with the call status, the available statuses are the following:

Answered

Status No Answer

Busy

Failed

Source Trunk
Select source trunk(s) and the CDR of calls going through inbound the trunk(s) will be filtered out.
Name

Destination
Select destination trunk(s) and the CDR of calls going outbound through the trunk(s) will be filtered out.
Trunk Name

Filter calls using the Action Type, the following actions are available:

Announce

Announcement page

Dial

Announcements

Callback

Call Forward

Conference

DISA

Follow Me

IVR

Page
Action Type
Parked Call

Queue

Ring Group

Transfer

VM

VMG

Video Conference

VQ_Callback

Wakeup

Emergency Call

Emergency Notify

SCA
Extension
Specify the Extension Group name to filter with.
Group

Select the fields that will be exported, the following fields are available:

Account Code

Session

Premier caller

Action type

Source trunk name

Destination trunk name

Caller number

Caller ID

Caller name

Callee number

Answer by

Context

Export File Start time

Data
Answer time

End time

Call time

Talk time

Source channel

Dest channel

Call status

Dest channel extension

Last app

Last data

AMAFLAGS

UIQUEID

Call type

NAT

Select the account Code to filter with. If pin group CDR is enabled, the call with pin group information will be displayed as part of the
Account Code
CDR under Account Code Field.

Specify the start time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the
Start Time
exact date and time.
 Specify the end time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the
End Time
exact date and time.

Enter the caller number to filter the CDR report. CDR with the matching caller number will be filtered out.

User could specify a particular caller number or enter a pattern. ‘.’ matches zero or more characters, only appears in the end. ‘X’

matches any digit from 0 to 9, case-insensitive, repeatable, only appears in the end.

Caller Number
For example:

3XXX: It will filter out CDR that having caller number with leading digit 3 and of 4 digits’ length.

3.: It will filter out CDR that having caller number with leading digit 3 and of any length.

Caller Name Enter the caller name to filter the CDR report. CDR with the matching caller name will be filtered out.

Enter the callee number to filter the CDR report. CDR with the matching callee number will be filtered out.

Callee Number
Note: The “Callee Number” filter field supports specifying Pattern (example: 3XXX) or using Leading digits (example: 3.) as filtering

options.

Table 147: CDR Filter Criteria

The call report will display as the following figure shows.

Figure 310: Call Report

The CDR report has the following data fields:

Start Time

Format: 2019-12-11 09:53:03

Action Type

Example:

IVR

DIAL

WAKEUP

Call From

Example format: 5555

Call To
Example format: 1000

Call Time

Format: 0:00:11

Talk Time

Format: 0:00:06

Account Code

Example format:

Grandstream/Test

Status

Answered, Busy, No answer or Failed.

Users could perform the following operations on the call report.

Sort by “Start Time”

Click on the header of the column to sort the report by “Start Time”. Clicking on “Start Time” again will reverse the order.

Download Searched Results

Click on “Download Search Result(s)” to export the records filtered out to a .csv file.

Download All Records

Click on “Download All Records” to export all the records to a .csv file.

Delete All

Click on button to remove all the call report information.

Delete Search Result

On the bottom of the page, click on button to remove CDR records that

appear on search results.

Note: When deleting CDR, a prompt will now appear asking whether to delete all recording files or not.

Play/Download/Delete Recording File (per entry)

If the entry has audio recording file for the call, the three icons on the rightest column will be activated for users to select. In the following picture, the second

entry has audio recording file for the call.

Click on to play the recording file; click on to download the recording file in .wav format; click on to delete the recording file (the call record

entry will not be deleted).

 Figure 311: Call Report Entry with Audio Recording File

Automatic Download CDR Records

User could configure the UCM630X to automatically download the CDR records and send the records to multiple Email recipients in a specific hour. Click on

“Automatic Download Settings” and configure the parameters in the dialog below.

Figure 312: Automatic Download Settings

To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By Week” or “By Month”, select Hour of the day as

well for the automatic download period. Make sure you have entered an Email or multiple email addresses where to receive the CDR records.

users have the option to delete the sent records “Delete Sent Records”

Starting from UCM630X firmware 1.0.10.x, transferred call will no longer be displayed as a separate call entry in CDR. It will display within call record in

the same entry. CDR new features can be found under Web GUI🡪CDR🡪CDR. The user can click on the option icon for a specific call log entry to view
details about this entry, such as premier caller and transferred call information.
 Figure 313: CDR Report

Figure 314: Detailed CDR Information

Downloaded CDR File

The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions.

Caller number, Callee number

“Caller number”: the caller ID.

“Callee number”: the callee ID.

If the “Source Channel” contains “DAHDI”, this means the call is from FXO/PSTN line.

Figure 315: Downloaded CDR File Sample

Context

There are different context values that might show up in the downloaded CDR file. The actual value can vary case by case. Here are some sample values and

their descriptions.

from-internal: internal extension makes outbound calls.

ext-did-XXXXX: inbound calls. It starts with “ext-did”, and “XXXXX” content varies case by case, which also relate to the order when the trunk is created.

ext-local: internal calls between local extensions.

Source Channel, Dest Channel

Sample 1:

Figure 316: Downloaded CDR File Sample – Source Channel and Dest Channel 1

DAHDI means it is an analog call, FXO or FXS.

For UCM6302, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.

For UCM6304, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports.
For UCM6308, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.

Sample 2:

Figure 317: Downloaded CDR File Sample – Source Channel and Dest Channel 2

“SIP” means it is a SIP call. There are three format:

(a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored.

(c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk.

(c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk. The last XXXXX is a random string and

can be ignored.

There are some other values, but these values are the application name which are used by the dialplan.

IAX2/NUM-XXXXXXX: it means this is an IAX call.

Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it.

Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases.

Playback: play some prompts to you, such as 183 response or run into an IVR.

ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA

Note: The language of column titles in exported CDR reports and statistics reports will be based on the UCM’s display language

CDR Export Customization

Users can select the data they want to see in exported CDR reports by first clicking on the Filter button on the CDR page under CDR🡪CDR and selecting the
desired information in the Export File Data field.
 Figure 318: CDR Export File data

CDR in GDMS Cloud

Cloud Storage for CDR Record which can be displayed under CDR 🡪 CDR in GDMS Cloud.

Figure 319: CDR in GDMS Cloud

Statistics

UCM supports the function of concurrent call statistics. This function provides users with statistics on the number of concurrent calls of all VOIP trunks (SIP

trunks and IAX trunks). Users can set search criteria to generate custom charts. Select the trunk and time to view the chart of the maximum number of

concurrent calls corresponding to the trunk in a certain day or month.

 Figure 320: CDR Statistics

Select one of the following trunk type.

All
Trunk Type
SIP Calls

PSTN Calls

Select one or more in the following checkboxes.

Inbound calls

Outbound calls
Call Type
Internal calls

External calls

All calls

By month (of the selected year).

By week (of the selected year).

Time Range By day (of the specified month for the year).

By hour (of the specified date).

By range. For example, 2016-01 To 2016-03.

Table 148: CDR Statistics Filter Criteria

Recording Files

This page lists all the recording files recorded by “Auto Record” per extension/ring group/call queue/trunk, or via feature code “Audio Mix Record”. If

external storage device is plugged in, for example, SD card or USB drive, the files are stored on the external storage. Otherwise, internal storage will be used

on the UCM630X.

Figure 321: CDR🡪Recording Files

Click on “Download” to batch-download the selected recording files.

Click on “Download All” to download all the recording files.

Click on “Delete” to batch-delete the selected recording files.

Click on “Clear” to delete all the recording files.

Click on “Scan” to retrieve the file information and display all the recording files on external storage. The UCM automatically retrieves the info of the

first 5000 files from external storage already. This button can be used when the number of files stored on the external storage exceeds 5000 files and it
requires manual file scanning.

Select either “USB Disk” or “Local” to show recording files stored on external or internal storage, depending on selected storage space.

Select whether to show call recordings, queue recordings or conference recordings.

Click on to download the recording file in .wav format.

Click on to delete the recording file.

To sort the recording file, click on the title “Caller”, “Callee” or “Call Time” for the corresponding column. Click on the title again can switch the sorting
mode between ascending order or descending order.

USER PORTAL
Users could log into their web GUI portal using the extension number and user password. When an extension is created in the UCM630X, the corresponding

user account for the extension is automatically created. The user portal allows access to a variety of features which include user information, extension

configuration and CDR as well as settings and managing Other features like Call Queue, Wakeup Service and CRM.

Users also can access their personal data files (call recordings, Voicemail Prompts …).
The login credentials are configured by Super Admin. The following figure shows the dialog of editing the account information by Super Admin. The

Username must be the extension number and it is not configurable, and the password is set on “User Password” field and it should not be confused with the
SIP extension password.

Figure 322: Edit User Information by Super Admin

The following screenshot shows an example of login page using extension number 1000 as the username.

Figure 323: User Portal Login

After login, the Web GUI display is shown as below.

 Figure 324: User Portal Layout

After successful login, the user has the following three configuration tabs:

Basic Information
Under this menu, the user can configure and change his/her personal information including (first name, last name, password, email address, department…).

And they can also set and activate their extension features (presence status, call forward, DND ….) to be reflected on the UCM.

Also, the user can see from this menu the Call Details Records and search for specific ones along with the possibility to download the records on CSV format

for later usage.

Personal Data
Under this section, the user can access and manage their personal data files which includes (voicemail files, call recordings …) along with the possibility to

set Follow me feature to without requesting the Super admin to set the feature from admin account.

Other Features
On this section, the user has access to manage and use all rich features which includes.

+ If user is a member of call queue, they can check the queue’s activity from the “Call Queue” section.

+ Create and enable Wake Up service.

+ Enable and configure CRM connection to either SugarCRM or Salesforce.

For the configuration parameter information in each page, please refer to [Table 149: User Management🡪Create New User] for options in User

Portal🡪Basic Information🡪User Information page; please refer to [EXTENSIONS] for options in User Portal🡪Basic Information🡪Extension page;
please refer to [CDR] for User Portal🡪Basic Information🡪CDR page.

MAINTENANCE

User Management
User management is on Web GUI🡪Maintenance🡪User Management page. User could create multiple accounts for different administrators to log in the

UCM630X Web GUI. Additionally, the system will automatically create user accounts along with creating new extensions for extension users to login to the
Web GUI using their extension number and password. All existing user accounts for Web GUI login will be displayed on User Management page as shown in
the following figure.

Figure 325: User Management Page Display

User Information

When logged in as Super Admin, click on ”Add” to create a new account for Web GUI user. The following dialog will prompt. Configure the parameters as

shown in below table.

Figure 326: Create New User

Configure a username to identify the user which will be required in Web GUI login. Letters, digits, and underscore are allowed
Username
in the username.

User Password Configure a password for this user which will be required in Web GUI login. English input is allowed without space,’ and “.

This is the role of the Web GUI user. When super admin creates new user, “Adminstrator” or customized privilege can be
Privilege
selected.

Multi-Factor If this authentication is enabled, the user account needs to be verified with an MFA code every time it logs in to enhance the

Authentication security of the product.

Email Address Configure the email address for the user. This is optional.
 Table 149: User Management🡪Create New User

When the super admin creates new user, the email address for the new user is optional. However, when the admin user created by super admin logs in to edit

user information, email address is mandatory. This email address is the same and will be sync up with the email address configured in login settings.

Once created, the Super Admin can edit the users by clicking on or delete the user by clicking on .

Figure 327: User Management – New Users

Multi-Factor Authentication

To enhance the security for UCM, super admin and admin can select to use multi-factor authentication method for login to protect the login information.

Virtual MFA and hardware MFA are supported and can be selected. Once enabled, the user must use MFA code to verify before login.

Note:

The user cannot enable or disable MFA for another different user.

Super admin can edit user settings for admin but cannot edit Multi-Factor Authentication option. MFA option is only viewable for super admin when

super admin edits other users.

When the user sees MFA enabled, only this user can disable or enable it again.

Email address and email settings are required before enabling Multi-Factor Authentication. Please ensure email setting has “Client” type configured.

Otherwise, MFA cannot be enabled.

Please refer to MFA how to guide here for more information.

Figure 328: MFA Settings

Custom Privilege

Four privilege levels are supported:

Super Administrator
 This is the highest privilege. Super Admin can access all pages on UCM630X Web GUI, change configuration for all options and execute all the

operations.

Super Admin can create, edit, and delete one or more users with “Admin” privilege

Super Admin can edit and delete one or more users with “Consumer” privilege

Super Admin can view operation logs generated by all users.

By default, the user account “admin” is configured with “Super Admin” privilege and it is the only user with “Super Admin” privilege. The Username

and Privilege level cannot be changed or deleted.

Super Admin could change its own login password on Web GUI🡪Maintenance🡪Login Settings page.

Super Admin could view operations done by all the users in Web GUI🡪Maintenance🡪User Management🡪Operation Log

Administrator

Users with “Admin” privilege can only be created by “Super Admin” user.

“Admin” privilege users are not allowed to access the following pages:

Maintenance🡪Upgrade

Maintenance🡪Cleaner

Maintenance🡪Reset/Reboot

Settings🡪User Management🡪Operation Log

“Admin” privilege users cannot create new users for login.

Note: By default, administrator accounts are not allowed to access backup menu, but this can be assigned to them by editing the option “Maintenance 🡪

User Management 🡪 Custom Privilege” then press to edit the “Admin” account and include backup operation permission for these types of users.

Figure 329: Assign Backup permission to “Admin” users

Consumer

A user account for Web GUI login is created automatically by the system when a new extension is created.

The user could log in the Web GUI with the extension number and password to access user information, extension configuration, CDR of that extension,
personal data, and Other Features. For more details; please refer to https://documentation.grandstream.com/knowledge-base/user-portal/.

The SuperAdmin user can click on on the “General_User” in order to enable/disable the custom privilege from deleting their own recording files,
changing SIP credentials, and disabling voicemail service in their user portal account.
 Figure 330: General User

Custom Privilege

The Super Admin user can create users with different privileges. 33 items are available for privilege customization.

API Configuration

Backup

Callback

Call Queue

Queue Statistics

Queue Recordings

CDR Recording Files

CDR Records

CDR Statistics

Dial By Name

DISA

Emergency Calls

Event List

Extensions

Outbound Routes

Inbound Routes

Fax/T.38

Feature Codes

IVR

Paging/Intercom

Parking Lot

Pickup Groups

PMS – Wakeup Service

Ring Groups

SCA
 Speed Dial

System Status

System Events

LDAP Server

Time Settings

Meeting

Voicemail

Voice Prompt

Wakeup Service

Zero Config

Announcement.

UCM RemoteConnect

Log in UCM630X as super admin and go to Maintenance🡪User Management🡪Custom Privilege, create privilege with customized available modules.

When you add CDR Records and CDR Recording Files custom privileges, additional privileges will appear (All Deletion of CDR and Allow Deletion of DCR

Recordings , respectively). This offers more flexibility on the privileges that the admin assigns to the user.

Figure 331: Create New Custom Privilege

To assign custom privilege to a sub-admin, navigate to UCM Web GUI🡪Maintenance🡪User Management🡪User Information🡪Create New User/Edit
Users, select the custom privilege from “Privilege” option.

Concurrent Multi-User Login

When there are multiple Web GUI users created, concurrent multi-user login is supported on the UCM630X. Multiple users could edit options and have

configurations take effect simultaneously. However, if different users are editing the same option or making the same operation (by clicking on “Apply

Changes”), a prompt will pop up as shown in the following figure.

 Figure 332: Multiple User Operation Error Prompt

User Portal/Wave Privilege

The user can create customize privileges related to an extension’s User Portal and Wave. The created privilege can be affected to the extensions to limit or

allow them to use certain functionalities related to Wave and the User Portal.

User Endpoint Access History

The User Endpoint Access History tab allows the administrator to view the access history of all extensions, the time on which the access has occurred, the IP

addresses from which the extensions were accessed, and whether they were accessed from the User Portal, Wave Web/Desktop, or mobile. Extension access
from the SIP endpoints won’t be logged in this page.
Login Settings

Change Password / Email

Change Password

After logging in the UCM630X Web GUI for the first time, it is highly recommended for users to change the default password to a more complicated

password for security purpose. Follow the steps below to change the Web GUI access password.

1. Go to Web GUI🡪Maintenance🡪Login Settings🡪Change Password / Email page.

2. Enter the old password first.

3. Enter the new password and re-type the new password to confirm. The new password has to be at least 4 characters. The maximum length of the

password is 30 characters.

4. Configure the Email Address that is used when login credential is lost.

5. Click on “Save” and the user will be automatically logged out.

6. Once the web page comes back to the login page again, enter the username “admin” and the new password to login.
 Figure 343: Change Password

Enter Old Password Enter the Old Password for UCM630X

Change Password Enable Change Password

Enter New Password Enter the New Password for UCM630X

Re-enter New Password Retype the New Password for UCM630X

Change Username Enable Change Username

Please enter the username Enter the Username

Email Address The Email address is the User Email Address. It is used for receiving password information if the user forgets his password.

Change Username

UCM630X allows users now to change Super Administrator username.

Figure 344: Change Username

Change Email Address

UCM630X allows user to configure binding email in case login password is lost. UCM630X login credential will be sent to the designated email address. The

feature can be found under Web GUI🡪 Maintenance🡪Login Settings🡪Change Password / Email
 Figure 345: Change Binding Email

Email Address Email Address is used to retrieve password when password is lost

Table 152: Change Binding Email option

Login Security

After the user logs in the UCM630X Web GUI, the user will be automatically logged out after certain timeout, or he/she can be banned for a specific period if

the login timeout is exceeded. Those values can be specified under UCM630X web GUI🡪Maintenance🡪Login Settings🡪Login Security page.

The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user does not make any operation on Web GUI within the timeout,
the user will be logged out automatically. After that, the Web GUI will be redirected to the login page and the user will need to enter username and password

to log in.

If set to 0, there is no timeout for the Web GUI login session and the user will not be automatically logged out.

“Maximum number of login attempts” can prevent the UCM630X from brutal force decryption, if this number is exceeded user IP address will be banned

from accessing the UCM for a period of time based on user configuration, the default value is 5.

“User ban period” specify the period of time in minutes an IP will be banned from accessing the UCM if the User max number of try login is exceeded, the

default value is 5.

“Login Banned User List” show the list of IPs’ banned from the UCM.

“Login Whitelist” User can add a list of IPs’ to avoid the above restriction, thus, they can exceed the User max number of try login.

Figure 346: Login Timeout Settings

Remote Login

This feature allows the user to manage trusted login locations, also, verifying where login sessions were initiated from, this is very important since, in this

type of scenario, the UCM6300 would be directly connected to the Internet, and the public IP address would be used for the remote login. This feature adds a
layer of visibility and control, thus enhacing the security of the UCM.

In this tab there are two types of lists of locations:

 Trusted Login Locations: These are the trusted login locations that are added manually by the admin. Any added trusted login location will not generate

any remote login alert upon the first time login.

Other Login Location: This list will show all the remote login locations that are not trusted, logging in for the first time from an untrusted login location

will generate an alert, but the subsequent remote logins from the same location will not generate alerts.

Figure : Remote Login

To add a trusted login location, the user must click on

Then add the public IP address of the location, click on “Check Location” to verify if it’s the correct location then click “OK“.

Note

The system administrator can add up to 5 Trusted Login Locations, while Other Login Locations can have an unlimited number of entries.

Operation Log
Super Admin has the authority to view operation logs on UCM630X Web GUI🡪Settings🡪User Management🡪Operation Log page. Operation logs list

operations done by all the Web GUI users, for example, Web GUI login, creating trunk, creating outbound rule etc. There are 7 columns to record the
operation details “Date”, “Username”, “IP Address”, “Results”, “Page Operation”, “Specific Operation” and “Remark”.
 Figure 347: Operation Logs

The operation log can be sorted and filtered for easy access. Click on or at the top of each column to sort. For example, clicking on for “Date” will
sort the logs according to newer operation date and time. Clicking on for “Date” will reverse the order.

Date The date and time when the operation is executed.

Username The username of the user who peformed the opertation

IP Address The IP address and geographical location from which the operation has been made.

Results The result of the operation.

Page Operation The page where the operation is made. For example, login, logout, delete user, create trunk and etc.

Specific Operation Click on the hyperlinked operation detail to reveal more details.

Remark Allows users to add notes and remarks to each operation.

User could also filter the operation logs by time condition, IP address and/or username. Configure these conditions and then click on ”Display Filter”.

Figure 348: Operation Logs Filter

The above figure shows an example that operations made by user “support” on device with IP 192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38

are filtered out and displayed.

To delete operation logs, users can perform filtering first and then click on to delete the filtered result of operation logs. Or users can

click on to delete all operation logs at once.

Upgrading
The UCM630X can be upgraded to a new firmware version locally. And in order to do that, please follow the below steps:

1. Download the latest UCM630X firmware file from the following link and save it in your PC.

http://www.grandstream.com/support/firmware

2. Log in the Web GUI as administrator in the PC.

3. Go to Web GUI🡪Maintenance🡪Upgrade, upload the firmware file by clicking on “choose file to upload” and select the firmware file from your PC.
The default firmware file name is UCM630Xfw.bin

Figure 349: Local Upgrade

Figure 350: Upgrading Firmware Files

4. Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

Figure 351: Reboot UCM630X

5. Click on “OK” to reboot the UCM630X and check the firmware version after it boots up.

Please do not interrupt or power cycle the UCM630X during upgrading process.

The firmware file name allows the use of the special characters besides the following restricted characters: # $ ^ & * + ( ) [ ] / ; ‘ | , < > ?

No Local Firmware Servers

Service providers should maintain their own firmware upgrade servers. For users who do not have TFTP/HTTP/HTTPS server, some free windows version

TFTP servers are available for download from

http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net

Please check our website at http://www.grandstream.com/support/firmware for latest firmware.

Instructions for local firmware upgrade via TFTP:

1. Unzip the firmware files and put all of them in the root directory of the TFTP server;

2. Connect the PC running the TFTP server and the UCM630X to the same LAN segment;

3. Launch the TFTP server and go to the File menu🡪Configure🡪Security to change the TFTP server’s default setting from “Receive Only” to “Transmit
Only” for the firmware upgrade;

4. Start the TFTP server and configure the TFTP server in the UCM630X web configuration interface;

5. Configure the Firmware Server Path to the IP address of the PC;

6. Update the changes and reboot the UCM630X.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server. Backup

The UCM630X configuration can be backed up locally or via network. The backup file will be used to restore the configuration on UCM630X when

necessary.

Backup/Restore

Users could backup the UCM630X configurations for restore purpose under Web GUI🡪Maintenance🡪Backup🡪Backup/Restore.

Click on ”Backup” to create a new backup file. Then the following dialog will show.

Figure 352: Create New Backup

1. Choose the type(s) of files to be included in the backup.

2. Choose where to store the backup file: USB Disk, SD Card, Local, NAS or GDMS.

3. Name the backup file.

4. Click on “Backup” to start backup.

Once the backup is done, the list of the backups will be displayed with date and time in the web page. Users can download , restore , or delete it

from the UCM630X internal storage or the external device.

Click on to upload backup file from the local device to UCM630X. The uploaded backup file will also be displayed in the web page and can be used

to restore the UCM630X.

 Users can restore backups of models with more FXO ports to models with less FXO ports as long as the configurations related to the extra FXO ports are removed.

Please make sure the FXO port settings, total number of extensions and total number of conference rooms are compactable before restoring to another UCM

model. Otherwise it will prompt a warning and stop the restore process as shown below:

Figure 353: Restore Warning

Figure 354: Backup / Restore

The option allows UCM to perform automatically backup on the user specified time. Regular backup file can only be stored in USB / SD

card / SFTP server. User is allowed to set backup time from 0-23 and how frequent the backup will be performed.
 Figure 355: Local Backup

Data Sync

Besides local backup, users could backup the voice records/voice mails/CDR in a daily basis to a remote server via SFTP protocol automatically under Web

GUI🡪Maintenance🡪Backup🡪Data Sync.

The client account supports special characters such as @ or “.” Allowing the use email address as SFTP accounts. It allows users as well to specify the

destination directory on SFTP server for backup file. If the directory does not exist on the destination, UCM630X will create the directory automatically
 Figure 356: Data Sync

Enable Data Sync Enable the auto data sync function. The default setting is “No”.

Account Enter the Account name on the SFTP backup server.

Password Enter the Password associate with the Account on the SFTP backup server.

Server Address Enter the SFTP server address.

Destination Specify the directory in SFTP server to keep the backup file. Format: ‘xxx/xxx/xxx’, If this directory does not exist, UCM will create
Directory this directory automatically.

Sync Time Enter 0-23 to specify the backup hour of the day.

Table 154: Data Sync Configuration

Before saving the configuration, users could click on . The UCM630X will then try connecting the server to make sure the server is up

and accessible for the UCM630X. Save the changes and all the backup logs will be listed on the web page. After data sync is configured, users could also

manually synchronize all data by clicking on instead of waiting for the backup time interval to come.

Restore Configuration from Backup File

To restore the configuration on the UCM630X from a backup file, users could go to Web GUI 🡪 Maintenance 🡪 Backup 🡪 Backup/Restore.

A list of previous configuration backups is displayed on the web page. Users could click on of the desired backup file and it will be restored to the
UCM630X.

If the backup was stored on GDMS, it will be displayed under Backups GDMS Cloud Storage, that can be restored by clicking on

If users have other backup files on PC to restore on the UCM630X, click on “Upload Backup File” first and select it from local PC to upload on the
UCM630X. Once the uploading is done, this backup file will be displayed in the list of previous configuration backups for restore purpose. Click on

to restore from the backup file.

 Figure 357: Restore UCM630X from Backup File

The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space in the file name.

The uploaded back file size must be under 10MB.

System Cleanup/Reset

Reset and Reboot

Users could perform reset and reboot under Web GUI🡪Maintenance🡪System Cleanup/Reset🡪Reset and Reboot.

To reboot the device, click on reboot icon.

To factory reset the device, click on reset icon, then all the configurations and data will be reset to factory default.

Figure 358: Reset and Reboot

User can also verify UCM certificate under the same path.
Cleaner

Users could configure to clean the Call Detail Report/Voice Records/Voice Mails etc… manually and automatically under Web GUI🡪Maintenance🡪System

Cleanup/Reset🡪Cleaner.

The following screenshot show the settings and parameters to configure the manual cleaner feature on UCM630X.

Figure 359: Manual Cleaning

UCM regularly cleans up CDRs, report data, chat data, recording files, historical appointment meeting records, voice mail, backup files, and fax files. The

report data includes queue statistics report and conference room call statistics report; chat data includes chat messages and chat shared files; historical

appointment conferences include audio and video conference appointment records. Automatic cleanup is not enabled by default and supports regular cleanup

of database data based on dimensions such as cleanup time, cleanup conditions, and cleanup interval.

User can also set an automatic cleaning under Cleaner🡪Automatic Cleaning. The following screenshot show the settings and parameters to configure the
cleaner feature on UCM630X.
 Figure 360: Automatic Cleaning

Enable

CDR Enable the CDR Cleaner function.

Cleaner

CDR Clean
Enter 0-23 to specify the hour of the day to clean up CDR.
Time
 By Schedule: If the clean interval is 3, cleaning will be performed every 3

days to remove all records that were generated 3 days ago.

Keep Last X Records: If the max number of CDR has been reached, CDR

will be deleted starting with the oldest entry at the configured cleaning
Cleaning

Conditions
time.(Note: The amount of records displayed on the page of call queue

statistics is not one-to-one with the actual amount of records in the

database.)

Keep Last X Days: Delete all entries older than X days.

Enter 1-30 to specify the day of the month to clean up CDR when By
Clean

Interval
Schedule is selected as Cleaning Conditions.

Set the maximum number of CDR entries to keep when Keep Last X
Max Entries
Records is selected as Cleaning Conditions.

Enter the number of days of call log entries to keep when Keep Last X days
Keep Last X

Day
is selected as Cleaning Conditions.

Enable
Report Enable scheduled queue log cleaning. By default, is disabled.

Cleaner

Queue Statistics Report

Cleanup Meeting Call Statistics Report

Type Scheduled Audio Meeting History

Scheduled Video Meeting History

Clean Time Enter the hour of the day to start the cleaning. The valid range is 0-23.

By Schedule: If the clean interval is 3, cleaning will be performed every 3

days to remove all records that were generated 3 days ago.

Cleaning
Keep Last X Records: If the max number of Queue Statistics has been reached, Queue Statistics will be deleted starting with the oldest
Conditions
entry at the configured cleaning time.(Note: The amount of records displayed on the page of call queue statistics is not one-to-one with the

actual amount of records in the database.)

Keep Last X Days: Delete all entries older than X days.

 Enter how often (in days) to clean queue logs when By Schedule is
Clean

Interval
selected as Cleaning Conditions. The valid range is 1-30.

Max Entries Set the maximum number of Queue Statistics entries to keep when Keep Last X Records is selected as Cleaning Conditions.

Enter the number of days of call log entries to keep when Keep Last X days
Keep Last X

Day
is selected as Cleaning Conditions.

Enable
Conference

Statistics
Enable scheduled Conference log cleaning. By default, is disabled.

Report

Cleaner

By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to remove all records that were generated 3 days ago.

Keep Last X Records: If the max number of Conference Statistics Report has been reached, Conference Statistics Report will be
Cleaning
deleted starting with the oldest entry at the configured cleaning time. (Note: The amount of records displayed on the page of call queue
Conditions
statistics is not one-to-one with the actual amount of records in the database.)

Keep Last X Days: Delete all entries older than X days.

Enter how often (in days) to clean queue logs when By Schedule is
Clean

Interval
selected as Cleaning Conditions. The valid range is 1-30.

Max Entries Set the maximum number of Conference Statistics Report entries to keep when Keep Last X Records is selected as Cleaning Conditions.

Enter the number of days of call log entries to keep when Keep Last X days
Keep Last X

Day
is selected as Cleaning Conditions.

Enable File
Enter the Voice Records Cleaner function.
Cleaner

Clean Files
If enabled the files in external device (USB/SD card) will be atomically
in External

cleaned up as configured.
Device
 Select the files for system automatic clean.

Basic Call Recording Files.

Choose Conference Recording Files.

Cleaner File
Call Queue Recording Files.

Voicemail Files.

Backup Files.

Clean time Enter the hour of the day to start the cleaning. The valid range is 0-23.

By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to delete all files.

Cleaning By Threshold: Check at the configured cleaning time every day to see if the storage threshold has been exceeded and perform

Conditions cleaning of all files if it has.

Keep Last X Days: Delete all files older than X days.

File Clean
Enter 1-30 to specify the day of the month to clean up the files.
Interval

Enter the internal storage disk usage threshold (in percent). Once this

File Clean
threshold is exceeded, the file cleanup will proceed as scheduled. Valid
Threshold

range is 0-99.

Automatically delete all recordings older than this x days when the threshold
Keep Last X

Days
is reached. If not set, all data is cleared

Cleaner Log Press Clean “button” to clean the cleaner’s log.

Table 155: Automatic Cleaning Configuration

All the cleaner logs will be listed on the bottom of the page.

USB/SD Card Files Cleanup

Users could configure to clean or download the Call Detail Report/Voice Records/Voice Mails automatically under Web GUI🡪Maintenance🡪System
Cleanup/Reset🡪USB / SD Card Files Cleanup.
 Figure 361: USB/SD Card Files Cleanup

Current Path Displays the current path.

Directory Select the directory user want to clean.

Delete Selected File Select multiple entries to delete from USB or SD card.

Table 156: USB/SD Card Files Cleanup

System Recovery
In some cases (for example after wrong upgrading procedure where the user doesn’t follow the correct steps to perform an upgrade) the system may go into

some hardware/software issues where the web UI access is lost as well as SSH, in this case the only solution would be to perform a full system recovery in
order to reset or update the software version of the device in order to use it again.

1. To access recovery mode on UCM, please follow below steps:

2. Remove the power from the unit and keep the network cable connected.

3. Press using a PIN the reset button and keep holding.

4. Plug back the power supply while maintaining the reset button pressed.

5. Wait for couple of seconds until you hear a click sound.

6. Release the reset button, and the system should display on the LCD a message “Recovery Mode” along with an IP address.

Once at this stage, the administrator can access the recovery mode web portal by typing in either the IP0 address (typically WAN) or IP1 address (typically
LAN) into a browser address bar. The following page should appear:
 Figure 362: UCM6302 Recovery Web Page

Make sure to enter the correct admin password, and press login to access the recovery mode page :

Figure 363: Recovery Mode

From here, the user can either upload a firmware file, factory reset or just reboot the device.

Syslog
On the UCM630X, users could dump the syslog information to a remote server under Web GUI🡪Maintenance🡪Syslog. Enter the syslog server hostname or
IP address and select the module/level for the syslog information as well as Process Log Level.

The default syslog level for all modules is “error”, which is recommended in your UCM630X settings because it can be helpful to locate the issues when

errors happen.

Some typical modules for UCM630X functions are as follows and users can turn on “NOTICE” and “VERBOSE” levels besides “error” level.

pbx: This module is related to general PBX functions.

pjsip: This module is related to SIP calls.

chan_dahdi: This module is related to analog calls (FXO/FXS).

Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is not recommended for daily usage. Too many
syslog prints might cause traffic and affect system performance.

The reserved size for Syslog entries on the cache memory of the UCM is 50M, once this sized is reached the UCM will clean up 2M of the oldest Syslog

entries to allow to save new logs.

Network Troubleshooting
On the UCM630X, users could capture traces, ping remote host and traceroute remote host for troubleshooting purpose under Web

GUI🡪Maintenance🡪Network Troubleshooting.

The following sections shows the steps to capture different types of traffic traces for analysis purposes.

Ethernet Capture

Ethernet Capture allows capturing the traffic of the UCM for troubleshooting purposes. To access Ethernet Capture feature, please navigate to Maintenance

→ Network Troubleshooting → Ethernet Capture

Figure 364: Ethernet Capture

The capture packets can be stored locally and downloaded for analysis. However, if the user is diagnosing a randomly-occurring issue, he/she can run a

continuous packet capture which can be limited by the size of the packet capture and the number of packet capture instances

Important

When the maximum packet capture file size is reached, a new packet capture file will be created. When the maximum number of capture files number is reached,

then the UCM will delete the oldest file created file and replace it with the new one.

Capture Type Ethernet Capture: Gets a packet capture of all network traffic going through the device.
 WebSocket Capture: Gets a packet capture of WebSocket protocol. Mainly used for troubleshooting Wave Web calling and

conferencing issues.

Interface Type Select the network interface to monitor.

Capture Filter Enter the filter to obtain the specific types of traffic, such as (host, src, dst, net, proto…).

● Local: Store the captured packets in the local storage.

● SFTP Server: Save the capture trace to a SFTP server. Please make sure that SFTP is correctly configured under PBX
Storage Location Settings -> Online Storage -> SFTP Server

● External Storage: Save the capture trace in a usb flash drive or an SD card. This requires that a USB flash drive or SD
card to be plugged into the UCM. File formats supported are FAT32 and ExFat.

When or more external storage units are connected to the UCM6300 series, the user will be able to pick which one to use.
Save to External Storage
Note: This option is available only when you choose "External Storage" as the storage destination of the capture trace.

When SFTP is selected, this option will appear. Please enter the directory path in which you would like to store the captured
Destination Directory
packets.

This option appears only when "External Storage" or "SFTP" options are selected.
Packet Capture Size
Define the packet capture size, the option available are: 50MB, 100MB, and 200MB.

Number of Packet Capture Define the maximum number of the packets captured. The available options are 5, 10, and 20 packets.

Start Start capturing network traffic.

Stop Stop capturing network traffic.

Download Download the captured packets. This option can only be used when the captured packets are stored locally.

Enable SRTP Debugging Check this box to troubleshoot calls encrypted with TLS/SRTP.

The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol,
port, port range) before starting to capture the trace.

Capture files saved on external devices will now have “capture” prepended to file names.

IP Ping

Enter the target host in host name or IP address. Then press “Start” button. The output result will dynamically display in the window below.
 Figure 365: Ping

Traceroute

Enter the target host in host name or IP address. Then press “Start” button. The output result will dynamically display in the window below.

Figure 366: Traceroute

Record Meeting for Diagnosis

Enter the target meeting, support the ongoing meeting, and then click the “Start” button to capture the recording diagnosis of the meeting members in
progress. The output result will be automatically displayed below, click the “download” button to download to the local. After the download is complete,

immediately click the “Delete” button to clear the system content.

 Figure 367: Record Meeting for Diagnosis

Signaling Troubleshooting

Analog Record Trace

Analog Record Trace

Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM630X user has caller ID issue for incoming call from Analog trunk.

Users can access analog record trance under Web GUI🡪Maintenance🡪Signal Troubleshooting.

Here is the step to capture trace:

1. Select FXO or FXS for “Record Ports”. If the issue happens on FXO 1, select FXO port 1 to record the trace.

2. Click on “Start”.

3. Make a call via the analog port that has the issue.

4. Once done, click on “Stop”.

5. Click on “Download” to download the analog record trace.

Figure 368: Troubleshooting Analog Trunks

A key Dial-up FXO

Users can directly set a PSTN number on the “External Extension” text box to troubleshoot issues related to the analog trunk easily, the following steps
shows how to use this feature:
 1. Configure analog trunk on UCM, including outbound route.

2. Enter a reachable external number in “External Extension”.

3. Press “Start” button. The call will be initiated to the external number.

4. Answer and finish the call before pressing “Stop” button.

The trace will be available for analysis to download after output result shows “Done! Click on Download to download the captured packets”.

Figure 369: A Key Dial-up FXO

When using a Key Dial-up FXO feature the outbound trunk for the analog trunk need to have internal permission. As well as it should be the trunk with the highest
outbound route priority.

After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream

Technical support in the following link for further assistance if the issue is not resolved.

https://helpdesk.grandstream.com/

Service Check
Enable Service Check to periodically check UCM630X. Check Cycle is configurable in seconds and the default setting is 60 sec. Check Times is the
maximum number of failed checks before restart the UCM630X. The default setting is 3. If there is no response from UCM630X after 3 attempts (default) to

check, current status will be stored and the internal service in UCM630X will be restarted.

Figure 370: Service Check

Network Status
In UCM630X Web GUI🡪System Status🡪Network Status, the users can view active Internet connections. This information can be used to troubleshoot
connection issue between UCM630X and other services.
 Figure 371: Network Status

APPENDIX A: RFC STANDARDS USED IN UCM6300 SERIES

RFC 3261 SIP: Session Initiation Protocol

RFC 3262 Reliability of Provisional Responses in SIP

RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers

RFC 3264 An Offer/Answer Model with the Session Description Protocol

RFC 3515 The Session Initiation Protocol (SIP) Refer Method

RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method

RFC 4028 Session Timers in the Session Initiation Protocol (SIP)

RFC 2976 The SIP INFO Method

RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

RFC 3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging

RFC 4733 RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals

RFC 4566 SDP: Session Description Protocol

RFC 2617 HTTP Authentication; Basic and Digest Access Authentication

RFC 3856 A Presence Event Package for the Session Initiation Protocol (SIP)

RFC 3711 The Secure Real-time Transport Protocol (SRTP)

RFC 4582 The Binary Floor Control Protocol (BFCP)

RFC 4583 Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams

RFC 5245 Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols

RFC 5389 Session Traversal Utilities for NAT (STUN)

RFC 5766 Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)

RFC 6347 Datagram Transport Layer Security Version 1.2

RFC 6455 The WebSocket Protocol

RFC 8860 Sending Multiple Types of Media in a Single RTP Session

 RFC 4734 Definition of Events for Modem, Fax, and Text Telephony Signals

RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples

RFC 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)

RFC 3550 RTP: A Transport Protocol for Real-Time Applications

CHANGELOG
This section documents significant changes from previous versions of the UCM630X user manuals. Only major new features or major document updates are

listed here. Minor updates for corrections or editing are not documented here.

Firmware Version 1.0.21.9

Added SMS service support. [SMS Settings]

Added support for creating shared departments. [Department Management]

Added support for RADIUS login. [RADIUS]

Added support for private intercom. [Configure Private Intercom]

Added support for setting an extension as the default destination in Click2call. [Integrated Customer Service]

Added support for SIP TLS cipher suite. [TCP & TLS]

Added size and packet number limit for continuous packet capture when using USB/SD card storage or SFTP. [Ethernet Capture]

Added support for resetting TLS certifcates to the default ones. [TCP & TLS]

Added support for SSH token. [SSH Access]

Added support for subscribing to a voicemail group. [Configure Voicemail Group]

Added support for setting separate call forwarding conditions for external and internal calls. [Create New SIP Extension]

Added support for forwarding calls to a custom prompt. [Create New SIP Extension]

Added support for external numbers to opt out of being recorded when calling into the UCM. [PBX Settings]

Added support for caller name look up. [Inbound Routes]

The name of the agent will now be displayed in the switchboard. [Switchboard]

Agent pause can now be performed quickly by dialing the respective feature code and the corresponding pause reason without having to interact with the
IVR. [Configure Call Queue]

Agent pause reason will now be displayed in the switchboard. [Switchboard]

Added support for enabling the welcome prompt to be played simultaneously with background music while the agent phone is ringing. [Configure Call
Queue]

Firmware Version 1.0.19.10

Specific time configuration is now included in extension exports

Firmware Version 1.0.19.9

Optimized various system processes

Added Onsite Meetings feature. [Onsite Meeting]

Added ability to customize extension call waiting tone [General Call Prompt Tones]
Updated Zoho CRM authentication process [Zoho Telephony]

{VM_DATE} date value format has been changed to MM/dd/yyyy hh:mm:ss from DDD yyyy MMM hh:mm:ss. [Configure Fax/T.38] [Voicemail Email
Settings]

Added Device Name ${DEVICE_NAME} variable to Alert Events and Emergency Calls email templates

Added Geolocation header support [Emergency Location Mapping]

Added P-Called-Party-ID header option to the Add/Edit Extension -> Features page [Create New SIP Extension]

Added Allow Operator Panel Monitoring extension option to toggle whether the Operator Panel can monitor the extension. [Create New SIP Extension]

Added Basic extension export option [Export Extensions]

Allow Call-barging Extension List option changed to Call Monitoring Whitelist [Create New SIP Extensions] [Create New IAX Extension]

Added Silence Suppression option to Extensions/VoIP Trunks page [Create New SIP Extension] [VoIP Trunk Configuration]

If a storage device is full, the UCM will mark it as unavailable and automatically change file storage path to the next available location based on the

Storage Path Priority. Previously, UCM would change the file storage path to its own local storage if external storage was full. [File Manager]

Added new commands related to call queue and Wave [API Configuration Parameters]

Added support for multiple API (new) users [API Configuration Parameters]

Added the Default Certificate Auto Renewal option. If enabled, the default browser certificate will be automatically renewed after 398 days (the max
certificate validity period of Chrome, Firefox, and Safari browsers). User-defined certificates are not affected. [HTTP Server]

Added ability to sync local IM data to Cloud IM [Cloud IM Service]

Added ability create custom IVR key presses [Custom Key Event]

Added Chat Data from Endpoint option to the Maintenance -> System Cleanup/Reset -> Cleaner page. If enabled, this option will clean out chat data

from Wave clients at the same time as the UCM’s server-side automatic/manual cleaning of chat data. [Cleaner]

Added support for meeting room passwords. However, meetings cannot be scheduled for rooms with meeting password enabled. [Multimedia Room

Configurations]

Meeting kick warning interval has been changed from 30 mins to 20 mins. Note: This kick warning will only play when there is only 1 person in a
meeting room, and if they do not opt to stay in the meeting room after the warning, they will be removed from the meeting room after 5 minutes.

ARP will now be used instead of ping to check NAS connectivity.

Upgrade logs will now contain firmware version information.

Queue chairmen can now log out dynamic agents

Added video paging support. [Configure 1-way Paging]

Added option to automatically reset user/Wave password upon check-in/check-out. [Local PMS]

Added option to clear Wave chat history automatically upon check-in or check-out. [Local PMS]

Added Local PMS functionality [Local PMS]

Check-out will no longer reset the “Skip Voicemail Password Verification” extension setting

Added ability to assign two extensions to a room [Room Management]

Added option to clear scheduled wakeup calls on both check-in and check-out. [Local PMS]

Added the ability to change the default call privilege of a room. A room’s privilege will be reset to this value after a guest checks out of it.

Added support for stereo audio recording [General Settings]

Added option to route calls based on a caller’s Diversion header value [Inbound Rule Configuration]
 Added ability to control whether to use failover trunks based on the call response codes [Failover Trunk Toggles]

Added support for H.264 with multiple payload types in SDP

When receiving an INVITE with no SDP, following INVITEs with SDP will offer H.264 1080p resolution by default.

In the scenario where an inbound external call is forwarded from an extension to an external number, the Contact header will now use the CID of the
forwarding extension instead of the caller’s CID.

Removed External Device Usage Threshold option. If a connected NAS has only 1GB remaining available storage space, it will be considered
unavailable and trigger the external disk usage alert. [NAS]

Added User Endpoint Access History page [User Endpoint Access History]

Added User Portal/Wave privilege control [User Portal/Wave Privilege]

Dial Trunk option has been renamed to Dial External Number and moved to the Dial Other Extensions section

The Wave Welcome email will now use the port number configured in System Settings->HTTP Server->Wave Settings->Port if the Wave Settings-

>External Host value is not a RemoteConnect address or does not contain a port number.

Added links to relevant online documentation to various pages of the UCM webUI.

Added Phonebook VMPK mode to GRP261x template

Added Firmware tab for improved firmware management [Firmware]

Added ability delete downloaded base model templates in the Model Update page

Added ability to search for templates via the device model name

Added ability to select either LAN1 or LAN2 to scan for devices on when using dual network method

Firmware Version 1.0.17.11

Several system process optimizations

Firmware Version 1.0.17.8

Updated python version to 3.8 and related processes.

Improved speed of applying changes

Updated lighttpd version to 1.4.61

The Privilege Name field now supports parenthesis ()

The Contacts page has been moved to its own category in the sidebar

Added https://www.zohoapis.in option to CRM Server Address list

Added Channel Path option for accessing specific IP camera channels via URL [Device Management]

Improved alert email sending process

Updated Remote Registration email template

Emergency calls will no longer be restricted by the RemoteConnect call limit.

The default highest priority codec is now G.722.

Added Remote Extension Privilege Update feature code and Remote Extension Privilege Update Whitelist field to allow specified users to remotely
change extensions’ privileges. [Feature Codes]

Automatic file migration after file storage path failure to the next storage location in the storage priority path.

Users can now customize the storage path priority for recordings, video recordings, and IM files. [File Manager]
FXO FSK CID detection now uses spandsp.

If SIP extensions synced from UCMs are deleted on GDMS, they will no longer be synced again.

If HA is enabled, the HA cluster IP address will now be provisioned as the config server to endpoints instead of the active UCM’s IP address.

Added support for configuring inbound route blacklist via HTTPS API.

If dialing into a Dial by Name directory, the call will end automatically after failing 3 times.

Added Server Type option to the LDAP Server→LDAP Phonebook→Phonebook Download Configurations page. Users can select between LDAP and

Active Directory. [LDAP Server]

Added Department field to LDAP phonebook contacts. [LDAP Server]

Added meeting room extensions to LDAP phonebooks. [LDAP Server]

External Contacts created from the Contacts page will now be added to the system’s internal LDAP phonebook. [LDAP Server]

Added Remote Login tab to the Maintenance→Login Settings page. [Maintenance]

Created new Meetings Settings page under the Multimedia Meetings page and moved several meeting-related options to it. [Multimedia Meeting]

Regular meeting participants can now invite other members to join the meeting by dialing 1 if “Allow User Invite” is enabled. [Multimedia Meeting]

Meetings will become “Pending” after rescheduling.

Pending meetings are now sorted by start time by default.

Added the Allowed to Override Host Mute option to the Edit Meeting Room and Schedule Meeting pages to allow participants to unmute themselves
even after the meeting host mutes them. [Multimedia Meeting]

Added support for user authentication. [OpenVPN]

Operation Log entries will now contain the IP address and location information from which the operation originated. [Operation Log]

Added option Automatically Clear Wakeup Calls for deleting scheduled wakeup calls after either guest check-in and check-out. [PMS Features]

Users can now dial the Update PMS Room Status feature code, the maid code, and the room status code all at once to change room status. [PMS

Features]

Added a Scan button to manually retrieve the list of recordings on external storage. The UCM automatically displays up to 5000 recordings on attached

external storage, but pressing this button will allow the UCM to display more. [Recordings Files]

Added ability to batch delete cloud storage files.

If IP endpoints cannot connect to the GDMS TURN server via UDP, UCM will use TCP to connect them.

Added Trunk Registration Period (s) option to SIP Settings->Misc. [SIP Settings]

Added option Special Attributes to the Extension/Trunk→VoIP Trunks→Edit SIP Trunk→Advanced Settings page. If enabled, the following attributes
will be included in the SIP SDP: ssrc, msid, mid, ct, as, tias, record. Enabling this may cause compatibility issues with non-Grandstream devices. [VoIP

Trunk Configuration]

Added CEI msid for audio calls.

Added trickle-ice param to SIP OPTION’s 200 OK.

profile-level-id will be added to 200 OK when receiving INVITEs without SDP.

Added ability to import/export speed dials. [Speed Dial]

Improved processes to avoid duplicate alerts for the following events: Registered SIP Trunk failed, Local Disk Usage and External Disk Usage.

Separated Allow Deletion of CDR and Recordings option to Allow Deletion of CDR and Allow Deletion of CDR Recordings. [User Management]

Added the Call Waiting option to the User Portal.

 Added support for voicemail message seeking. When listening to voicemail, users can press star (*) to rewind 3 seconds or pound (#) to fast forward 3

seconds. [Voicemail]

Added Line Selection Strategy option for Trunk Groups. [VoIP Trunks]

Changed register trunk Username field name to Trunk Registration Number. [VoIP Trunks]

General web UI improvements

Added Help option under the username dropdown menu that will redirect to the UCM6300 Series FAQ.

Updated some tooltips.

Optimized search functionality

Added GMT+2:00 (Israel) option to Time Zone drop down list in all Zero Config pages.

Added support for WP22 and WP825 model templates.

Firmware Version 1.0.15.13

Added SNMP monitoring feature. [SNMP]

Added support to configure the time of a holiday. [Holiday]

Added ability to determine the maximum total call duration per trunk for outbound calls.

Added contact viewing privilege (independent from Department Contact Privilege). [Contacts]

Added support for agent ID announcement. [Call Queue]

Added support for Service Level Agreement for Call Queue. [Service Level Agreement]

Added support for changing the Meeting room’s name.

Added WebRTC Trunk feature. [WebRTC Trunks]

Extension data cleaning has been improved. [Search and Edit an Extension]

Added support for SRTP Crypto Suite. [Create New SIP Extension]

STIR/SHAKEN has been improved. [STIR/SHAKEN]

Added support for displaying the extension that initiated an emergency call in the emergency email notification.

Added support for collecting ICE candidates when an RTP connection is requested. [RTP Settings]

Added support for recording a Multimedia Meeting shared screen only with the audio of the participants.

Flood Attacks and Network Traffic Storm alerts have been added to the Alert Events List.

Added support for Network Port Traffic Control for the ports of the UCM63xx.

Added Support for limiting the frequency of calls that can be made in a period of time.

Added support for storing the local chat files in the GDMS.

UCM RemoteConnect plan expiry screen has been improved.

GDMS Cloud Storage Space details can now be viewed in the RemoteConnect menu.

Firmware Version 1.0.13.9

Added option to enable/disable DND status remotely for an extension. [CALL FEATURES]

Added local proxy in IM settings. [Table 6: Cloud IM Parameters]

Added support for enabling/disabling auto audio/video recording for meeting. [Auto Record]
 Added privilege management for contacts. [Privilege Management]

Improved fail2ban blacklist display. [Blacklist]

Improved email template for scheduling meeting. [Email Templates]

Contacts sync-up between UCM and end points (wave/IP phones). [LDAP Settings]

Added ability to specify DOD number based on outbound route. [Outbound Routes DOD]

Fixed an issue where updating model templates will result in deleting the existing ones.

Added support to use TURN Relay as an option to allow hosts behind NAT firewalls to communicate. [VoIP Trunk Configuration]

Firmware Version 1.0.11.10

Combined audio meeting and video meeting to multi-media meeting. [MULTIMEDIA MEETING]

Updated UCM6301 public meeting room number from 3 to 4. [MULTIMEDIA MEETING]

Added operator panel. [OPERATOR PANEL]

Added time condition support for IVR key events. [Key Press Event]

Supported adding comments for fail2ban whitelist. [Fail2Ban Whitelist]

Added Call Flip feature code. [Call Flip]

Added cloud IM abnormal alert event. [Cloud IM abnormal]

Added option “Play Follow Me Prompt”. [Play Follow Me Prompt]

Add option “Ignore 180 Response”. [Ignore 180 Response]

Added Multi-Factor Authentication for UCM login. [Multi-Factor Authentication]

Added support for IoT device management. [DEVICE MANAGEMENT]

Added remark for UCM system status. [Remark]

Added option to enable and disable passwordless for remote access. [Figure 255: UCM RemoteConnect Plan Settings]

Added option to enable and disable virtual queue call back keys settings. [Virtual Queue Callback Key]

Added option to enable IPv6 for HA settings. [Enable IPv6]

Added Address Book Management section. [Address Book Management]

Added option “Stop Ringing”. [Stop Ringing]

Add option “Email Missed Call Log”. [Email Missed Call Log]

Add option “Security Mode” for NAS settings. [Security Mode]

Removed display for consumer users on the user management page. [Figure 326: Create New User]

Firmware Version 1.0.9.10

Added Support for import/export Zero Config. [Figure 68: Global Policy Categories]

Added support for Enable Wave and Sync Contact under the extension. [Enable Wave][Sync Contact]

Added support for Custom time supplement time conditions. [Time Condition]

Added support for Call Restriction. [RESTRICT CALLS]

Added support for Queue Metrics. [QUEUE METRICS]

 Added support for CDR API whitelist. [Permitted IP]

Added support for Call queue satisfaction survey. [Queue satisfaction statistics ][Agent satisfaction statistics].

The old API Configuration is reopened for use. [HTTPS API Settings (Old)]

Custom permissions support the function of deleting CDR and recording files. [delete CDR and recording files]

Added support to adjust the recording file storage path. [File Storage Management]

High Availability feature on UCM6300 series. [HA]

Paging/Intercom supports delayed paging. [Delayed Paging]

UCMRC remote service diagnosis. [Figure 253: Remote Diagnosis]

Support LDAP to automatically update the phone book. [LDAP Automatic Update Cycle]

Support meeting room automatic gain control. [Meeting AGC]

Firmware Version 1.0.7.12

Added support for an email reminder when editing the time of a scheduled meeting. [Email Reminder (m)]

Improved extension status syncing process to the IM server.

Firmware Version 1.0.7.9

Statistics and alarm settings of the maximum number of concurrent calls on the trunk. [Enable Concurrent Call Alert] [Enable Concurrent Call Alert]

Add ability to enable/disable Ring Group Voicemail and make Ring Group Voicemail have configurable voicemail prompts. [Voicemail]

Add ability to import/export pickup groups and time settings. [PICKUP GROUPS]

Add option to set a holiday as fixed. [Year]

Support DOD digit manipulation based on extension. [Direct Outward Dialing (DOD)]

Support to join the scheduled meeting 10 minutes in advance and is controlled by conference resources. [Meeting Schedule]

Clean up the history of audio and video schedule meetings regularly. [Cleanup Type]

Record the actual member information in the meeting history details. [Figure 151: Meeting details]

Support to initiate a random meeting in a scheduled/instant-meeting room. [Meeting Schedule]

The host password for the scheduled meeting is randomly generated. [Meeting Schedule]

Add the “Download Client” link to the extension email template and add the Wave promotion guide in the setup wizard. [Email Templates] [Setup

Wizard]

Added support for Integrated Customer Service SDK. [Integrated Customer Service SDK]

Support customer service system call (Click2Call) service provides SDK download for customers to integrate into the website and set up security defense
Fail2ban. [Customer Service System Call Defense]

Support to use Cloud IM service. [IM]

Add email reminder configuration in the audio meeting room. [Email Reminder (m)]

Added support for custom prompts under Voicemail Group. [Busy Prompt][Greet Prompt][Temporary Prompt][Unavailable Prompt]

The dashboard shows eMMC lifetime and the number of public/random video meetings. [PBX Status]

Added some System Alert Events. [Alert Events List]

Firmware Version 1.0.5.4

 Added support for v-Fax, Fax-sending, Email2Fax. [FAX SERVER]

Added ability to restore backups remotely stored on GDMS [Restore Configuration from Backup File]

Added support for STIR/SHAKEN. [SIP Settings/STIR/SHAKEN] [VoIP Trunk Configuration]

Added ability to restrict calls and features based on CPU usage and data partition usage. [General Settings]

Added NAT option to the Export File Data filtering option. [Export File Data]

Wave desktop client is now supported. [Wave Desktop]

Added support for IAX. [IAX SETTINGS] [EXTENSIONS] [VoIP Trunk Configuration]

Improved audio and video conference pages. A list of the meetings that have not started and a meeting history list have been added. [Meeting Schedule]

Support sending post-meeting reports to the host after scheduling a meeting. [Meeting Schedule]

Added the SRTP Debugging option to the Ethernet Capture page. [Enable SRTP Debugging]

Updated the System Cleanup/Reset page interface. [System Cleanup/Reset]

Added manual cleaning. [Cleaner]

Added ability to add a custom browser tab icon and custom logos on various pages of the web management portal and Wave Web portal. [UI
Customization]

Improved push notification for Wave mobile.

Added Instant Messaging functionality to Wave Web. [Wave Web]

LDAP phonebook information will now be synchronized when viewing the Contacts page. [Wave Web]

UCM can now synchronize system event alerts to GDMS. [Plan Settings]

Added threshold-based Call Control & Data Write Control. [General Settings]

Added Layer 3 QoS for SIP and Layer 3 QoS for RTP options to global policy and relevant templates. [Layer 3 QoS For RTP] [Layer 3 QoS For SIP]

Improved device list import support and added the ability to export devices on the ZeroConfig Device List page. [Managing Discovered Devices]

Firmware Version 1.0.3.10

Added RemoteConnect Mode option in SIP trunk. [RemoteConnect Mode]

Cloud Storage for CDR Backup and Record. [CDR in GDMS Cloud]

Add support for IDS PMS interface. [IDS PMS]

Add ability to enable PMS Wakeup call from remote extensions. [PMS Remote Wakeup Service]

Add ability to configure local country code for an outbound route. [Local Country Code]

Add support for Pattern and Leading digit filtering options in Callee Number. [Callee Number]

Added description of SSH switch options in LCD menu. [Use the LCD Menu]

Add the ability to import/export DOD. [Direct Outward Dialing (DOD)]

Add the ability to enable auto record per inbound/outbound route. [Auto Record] [Auto Record]

Add the ability to Enable Recording Whitelist. [Enable Recording Whitelist]

Add the ability to select missed call type to be sent via email. [Missed Call Type]

The Filter action is now supported in AMI sessions. [ASTERISK MANAGER INTERFACE]

Added command to delete call recordings after downloading them. [recapi]

 Added commands to add, edit, and delete PIN groups. [addPinSets] [getPinSets] [updatePinSets] [deletePinSets]

Added External Disk Status alert event for monitoring external storage connection status. [Alert Events List]

The language of column titles in exported CDR reports and statistics reports will now be based on the UCM’s display language. [Downloaded CDR File]

Added Parking Lot Timeout Alert-Info option. This will add the specified alert-info header value to parking timeout callbacks. [Parking Lot Timeout
Alert-Info]

Updated HMobile check-in request format. [HMobile PMS Connector]

Updated HMobile Mini Bar request. [HMobile PMS Connector]

Added several updates to HSC PMS. [HSC PMS]

Added the ability to specify the reason for agent pause (*83 by default). [Agent Pause]

Added CDR Stored in GDMS Cloud option for RemoteConnect Plan Settings. [UCM RemoteConnect]

GDMS Cloud Storage has been added as a recording storage location in the PBX. [PBX Settings/]

After adding a UCM to GDMS, its RemoteConnect address will automatically be added as a SIP server to the GDMS account. [UCM RemoteConnect]

Added the Ringback Tone option. Users can now select a custom prompt to play as a ringback tone for callers dialing in via the selected inbound route.
[Ringback tone]

Ring Group Voicemail can now be set as a routing destination and an IVR key press destination. [Default Destination]

Added the following custom privileges: LDAP Server, UCM RemoteConnect, and Announcement. [Custom Privilege]

Added a Forgot Password option to the Wave Web login page. [Wave Web]

Added a new command to set the same-day wakeup service by dialing *36. [Wakeup Service]

Firmware Version 1.0.2.25

This is the initial version.