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Valcom Engineered Solutions VoIP Gateways and Servers r 2022-1.10

Valcom IP systems are used in facilities worldwide. In addition to a full complement of PoE IP
speakers, horns and LED signs, there are a number of gateways and servers. Audio broadcast
to PoE IP speakers and horns may be initiated by:

1) Audio sources connected to Audio Gateways

2) Telephones or telephone systems connected to FXS Gateways (via DTMF)
3) FXS ports connected to FXO Gateways (via DTMF)
4) Session Initiation Protocol (SIP) using SIP Paging Gateways for group audio
5) Interactive Consoles
6) Application Servers

All Gateways are powered by PoE or include a power supply, and all servers include a power
supply.

There is no practical limit to the number of endpoints that may be included in a Valcom
IP6000 system. The system can be sized for any application!

Audio Gateways – VE8001B, VE8002B, VE8004B, VE8001BR, VE8002BR, VE8004BR

Audio Gateways are available in one, two or four port models. Each port can either provide an
audio output or an audio input. As an output, the channel provides an adjustable audio feed to
either an amplifier line level input or to self-amplified speakers.

As an input, each channel can be connected to a line level, monophonic audio source. The audio
inputs may be switched on and off via a contact closure, VOX or by an application server. The
audio may be directed to any group of speakers. If your audio source is stereo, a V-9130-W single
gang passive stereo to monophonic mixer may be used to mix the channels.

Audio Gateway channels may be directly accessed via Session Initiation Protocol (SIP).

FXS Gateways – VE801X

FXS Gateways are available in one, two or four port models. They provide dial tone for DTMF
access of channel dial codes and groups and may be connected to a POTs telephone (standard

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phone) or a telephone system input that is designed to connect to dial tone (FXO port, C.O. Line
Port, Loop Start Trunk Port). They cannot be connected to another dial tone source (station port,
telephone line).

In addition, VE801X can be a member of a paging group. In this capacity, upon joining the group,
the VE801X will ring the connected terminal device (phone, phone system port) and deliver the
message.

FXO Gateways – VE802X

FXO Gateways are available in one, two or four port models. They accept ring voltage and return
dial tone for DTMF access of channel dial codes and groups and may be connected to an analog
station port of a phone system. To ensure proper disconnect, the analog station port must be able
to provide Open Loop Disconnect.

In addition, VE802X can be a member of a paging group. In this capacity, upon joining the group,
the VE802X can auto dial a zone/group on an analog intercom/paging system and deliver the
message.

For the purpose of providing a SIP interface to analog Valcom systems, FXO Gateway channels
may be directly accessed via Session Initiation Protocol (SIP) and, if connected to a multi-zone
analog Valcom system, return Valcom dial tone for zone selection.

They can also route incoming, ringing FXS connections (such as those initiated by talkback
speaker call buttons) to SIP phones. Once answered, the audio connection is a standard full
duplex 2-way phone call connection. (Note that proper caller ID passage may require the use of
SIP Trunk mode).

SIP Paging Gateways – VE20X

SIP Paging Gateways are available in one or four port models. The purpose of the SIP Paging
Gateways, regardless of the number of ports, is to provide SIP access to paging groups. There
are two modes of operation in this regard. Users may access up to eight individual groups per
VE20X, or each VE20X may be registered as a SIP Trunk for access to up to 100 groups. If
utilized, each audio port requires one of the available SIP identities. The audio outputs are
functionally identical to VE800X audio outputs. The VE20X SIP Paging Gateways may also be
used to provide SIP invoked loud ringing.

SIP Intercom Controller – VE8090R

The VE8090R SIP Intercom Controller provides access to Valcom VoIP audio endpoints from SIP
(Session Initiation Protocol) telephone servers. The SIP Intercom Controller can communicate
with the SIP telephone server as either registered SIP stations or as a SIP trunk. The SIP
Intercom Controller supports up to four concurrent calls to either talkback intercom or one-way
group paging.

In Station mode, the VE8090R allows registration of up to four SIP Stations, each of which will
return dial tone and allow the caller to enter a Valcom dial code or group number. Station mode

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additionally supports IP endpoint (2-way speaker) call button auto destinations to a hunt group of
up to 4 SIP destinations.

In Trunk mode, the VE8090R provides support for up to four concurrent calls. The inbound phone
number is automatically interpreted to be a Valcom group or channel dial code and the VE8090R
attempts to call that number via Valcom’s protocol. As an added benefit, Trunk mode operation
processes the inbound SIP phone number and strips leading digits to match the dial code length
programmed into the Valcom system. For example. if the SIP PBX routes phone number
5405632000 to the VE8090, and the Valcom system is programmed to use a 4-digit dial code
length, then only the “2000” part of the number will be used to make the Valcom page or intercom
call.

Trunk mode also supports IP endpoint (2-way speaker) call button auto destination, but to a single
SIP destination.

The VE8090R also provides 2 convenient FXS ports to facilitate the addition of POTS telephones,
Loop Start Trunk or FXO ports.

V-9972 Universal Paging Interface

The V-9972 Universal Paging Interface is designed to provide access to analog paging systems
from an analog station port, (FXS, PABX), PABX loop start Trunk Port (FXO) or SIP (Session
Initiation Protocol) connection. It features a loop start output as well as a line level audio output.

The V-9972 is used:

1) to provide SIP or FXS access to an analog multizone one-way or talkback paging controller t/r

2) to provide SIP/FXO/FXS access to an IP paging system

3) to provide SIP/FXO/FXS access to an integrated line level output

It can route calls backwards (for example, from a call button) to an analog or SIP/IP destination.

The L9972-1 license expands SIP from 1 to 4 extensions, adds the ability to perform Night Ring
and adds Store & Play on all lines.

The L9972-2 license includes the L9972-1 license and enables the ability to input Valcom IP dial
codes (group or channel) to work directly with IP groups or IP speakers, etc.

The V-9972-2 is the V-9972 with the L9972-1/L9972-2 licenses pre-installed.

The 9972 can also listen to Valcom page groups and play audio from its integrated One-way Page
Out connection (much like an Audio Gateway).

VEUTM Unicast/Multicast Converter

The VEUTM converts multicast packets to and from unicast packets in order to transport multicast
traffic over a unicast only network.

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8-Port Gateways – VE8025/VE8045

8-Port Gateways are intended for upgrading existing analog intercoms. Each channel provides
an adjustable talkback audio feed to either one or two 45-ohm (VE8045) or up to a 5-watt load of
old fashioned 25-volt (VE8025) interior intercom speakers. Two of these channels are dual mode
and feature a parallel line level output for direct connection to an amplifier line level input or to
self-amplified speakers. Additionally, one normally open call button input is available per channel.
An auxiliary input is provided for local program material. 8-Port Gateway channels may be directly
accessed via Session Initiation Protocol (SIP).

12-Port Talkback Gateway - VE1225

The VE1225 12 Port Talkback Gateway allows for the ultimate modernization of 45 Ohm or old
fashioned 25-volt intercom systems. Each VE1225 provides 12 two-way intercom channels. Each
channel may support up to up to 2 45-Ohm speakers, or a 15-watt load of 25-volt speakers. It
also features 12 call button inputs and dual line level outputs. These line level outputs may be
connected to amplifiers or self-amplified speakers to provide one-way audio to common areas.
AC power makes installation as easy as plugging in a common AC receptacle. System relays may
be utilized for visual signaling during announcements or for door unlock functionality. System
capacity is virtually unlimited. 12-Port Talkback Gateway channels may be directly accessed via
Session Initiation Protocol (SIP).

IP/SIP 20-Watt Paging Amplifiers – VIP-851-XX

IP/SIP 20-Watt Paging Amplifiers allow SIP (or Valcom Protocol) access of a single zone of old
fashioned 25/70.7/100 volt speakers or horns.

VENSCA Network Station Card Adapter

The VENSCA was developed to seamlessly bridge the gap between legacy Valcom analog station
cards and IP technology.

Each VENSCA can support up to eight compatible station cards of up to 24 zones/stations each.

Analog station cards connect to the VENSCA via a 40-pin ribbon cable. Audio distribution from
station cards to speakers is accomplished via analog signaling. Other Valcom IP endpoints may
be seamlessly integrated with VENSCA based systems and multiple VENSCA based systems
may be combined for multi-facility communications.

There is no practical limit to the number of VENSCAs in a system.

VE8006R Admin I/O Gateway

The VE8006R Admin I/O Gateway is an ideal interface for VENSCA based systems and provides
two administrative dial tones (admin ports), four programmable Form A relay contacts, two

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auxiliary audio inputs, and two programmable switch inputs. All the I/Os are accessible via a male
Amphenol connector.

VE8048A Networked Input & Relay Module

Each VE8048A provides eight latching or momentary switch input and eight normally open relay
outputs. The relay outputs may be used to provide control across the network. The relays may be
activated individually or as a group by:

a) one or more of the switch inputs

b) a VE6025 Application Server

The latching or momentary switch inputs may also be used to initiate action from an Application
Server (audio broadcast, text broadcast, launch an emergency message, etc.).

VE8092 Interactive Console

The VE8092 Interactive Console is a programmable touchscreen device that sends live and/or
pre-recorded announcements, pushes text notifications to LED signs, can make and receive calls
from Valcom and SIP endpoints, and receives audio pages.

VE8092 SPECIFICATIONS

Access Methods include Touchscreen activation of announcements, SIP protocol phone calls,
Valcom intercom and page group calls

Features include:

a) Desktop interactive IP device with 10.1” color touch screen display, built in beam-forming
microphone array, and loudspeaker.
b) Sends pre-recorded audio files at the push of a button.
c) Sends microphone pages, both “Live Mic” and Record & Send for feedback elimination
d) Sends text messages to Valcom LED signs
e) Makes/receives bidirectional voice calls to Valcom and SIP endpoints
f) Sends recorded audio pages
g) Receives audio pages
h) Simple configuration via web browser interface
i) Adjustable display tilt for glare reduction
j) Power over Ethernet Plus (PoE+), or external 24VDC power adapter

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VIP-MC Master Clock

The VIP-MC is a universal master clock that provides RS-485, BCD, NTP, Daytime as well as
most popular analog (relay driven) clock correction protocols.

VE6025 Application Server Pro

The main purpose of the VE6025 is to distribute audio and text files to groups of speakers or LED
signs. It is commonly used as a bell scheduler in schools or businesses. It stores 25000 seconds
of wav formatted audio files and can distribute them to groups manually or automatically based
upon schedules. It features multiple ways to add or record audio files and a graphical interface
for point and click message launch. The VE6025 also can receive and process emergency
messages from CAP sources. This ability automates the deployment of emergency information
throughout a facility. Message destinations are determined by the content of each individual CAP
message. A powerful on-board text-to-speech engine converts the text from the CAP message
into spoken language. The VE6025 can additionally send emergency text messages to both LED
signs and pop-up windows on networked PCs.

VE6023 Telephone Paging Server

Many modern facilities utilize IP phone systems. As part of a Valcom IP paging solution, the
VE6023 turns many IP telephones into small paging speakers. This allows the telephones to
provide paging coverage in area where speakers may not be present.

VE6024 eLaunch Server

The VE6024 eLaunch Server provides advanced “one click” emergency notification launch
capabilities. It allows users to quickly deploy emergency messages by e-mail, RSS feeds, ATOM
Feeds, HTTP pushes, CAP feeds and other. This “one click” deployment saves time and assures
consistent message delivery during stressful crisis situations. For an explanation of CAP visit
http://en.wikipedia.org/wiki/Common_Alerting_Protocol .

VE6030-1 Communication / Notification Server

The VE6030-1 is a combination of an Application Server (same as VE6025) and a Windows®

based PC on a 1U 19” rack shelf. Both the Application Server and Windows® based PC are
browser accessible.

VE4804 Aux Audio I/O Controller

The VE4804 is a combination of a VE8004B 4 port audio Gateway and a VE8048A I/O
Gateway on a 1U 19-inch rack shelf.

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VE6090-1 Communications/Notification Server with SIP and UPS

The VE6090 is a combination of a VE6030-1, a commercial UPS and a VE8090R SIP

Intercom Controller on a 1U 19-inch rack shelf.

VEIP6K-1 Communication & Notification Controller

The VEIP6K-1 is comprised of a VE4804 and a VE6090-1

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 Input to IP6000 System Device Type(s) Required Note
One VE20X per eight audio
groups, or one per 100 audio
groups in SIP trunk mode
VE20X SIP Paging Gateway or
SIP
VE8090R SIP Intercom Controller and/or

One VE8090R per 4

simultaneous talk paths
Text to IP LED Signs VE6025/VE6030-1 Server

Syslog Data VE6025/VE6030-1 Server Syslog Data or Contact Closures

VE8048A Network Relay/Input Module may be used to trigger text and
Contact Closure audible messages
VE6025/VE6030-1 Server
Text to PC Screens (pop up) VE6025/VE6030-1 Server
The VE6024 is commonly used
CAP (Common Alert Protocol) VE6025/VE6030-1 Server as a primary CAP source for the
VE6025
Audio must be monophonic
VE800X Audio Gateway or VE8006R
Line level audio feed (i.e. music) A N.O. output contact is
Admin I/O Gateway
desirable1
VE800X Audio Gateway or VE8006R A N.O. output contact is
Phone system page port
Admin I/O Gateway desirable1
Microphone audio level must be
VE800X Audio Gateway or VE8006R boosted to line level.
Microphone
Admin I/O Gateway Microphone should have a N.O.
output contact1
VE801X FXS Gateway. VE8090R SIP
FXO Port Controller or VE8006R Admin I/O
Gateway
VE801X FXS Gateway. VE8090R SIP
POTS (plain old telephone set) Controller or VE8006R Admin I/O
Gateway
FXS Port VE802X FXO Gateway FXS port should provide OLD2

Any or all of these inputs may be utilized in a single system. For example, SIP with a POTS or
microphone backup in case of phone system failure.
1
The audio input of a VE800X audio gateway may be triggered by:

a) a N.O. (normally open) contact closure

b) a VE6025/VE6030-1 Server Event
c) VOX (voice operated switching), which depends upon constant audio. Breaks in the
audio will result in undesirable switching.
2
OLD is Open Loop Disconnect.

Once a connection is established, the VE802X FXO Gateway can disconnect from:

a) Open Loop Disconnect (best option)

b) detected silence (requires a quiet line)
c) a fixed timeout (which may interrupt the ongoing announcement)

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Valcom VoIP gateway audio outputs and Valcom VoIP speakers are each programmed with a
unique channel dial code. These channel dial codes allow users to broadcast to, and in some
cases, receive audio from, specific channels. The individual channel dial codes are then combined
in one or more groups to allow simultaneous one-way audio broadcasts.

Valcom servers only use group access. Gateways use channels and groups in various ways:

Can Send Audio Can Receive Audio

Endpoint
To Groups To Channels Group Audio Channel Audio
FXO Gateway Via Dial Code Via Dial Code Via Dial Code1 Via Dial Code3
FXS Gateway Via Dial Code Via Dial Code Via Dial Code2 From Channels3
SIP Intercom Controller Via Dial Code Via Dial Code Via Dial Code2 From Channels3
SIP Gateway Via SIP4 No8 Via Dial Code Via Dial Code5
Audio Gateway Via Input6 No8 Via Dial Code Via Dial Code
Application Server Yes No8 For recording No8
Telephone Paging Server Yes7 No8 Yes7 No8
To FXS/FXO
8/12 Port Gateways No Channels via Via Dial Code Via Dial Code
Call Button

1 FXO gateways that receive group audio access a peripheral system via a user defined DTMF dial
code. Once the peripheral system is connected, it receives the audio in progress. (See screenshot
below).

2
FXS gateways that receive group audio ring the connected terminal device (telephone, C.O. Line Port, FXO Port,
Loop Start Trunk Port) while broadcasting the group audio. Once answered, the terminal device joins the audio in
progress.

3
Via auto-destinated talkback speaker call buttons. This requires that the seized station port auto dial the desired
telephones.

4
Audio received via a SIP identity is rebroadcast to selected Valcom groups.

5
Requires origination from a FXO or FXS gateway. For SIP access the channel must be added to a dedicated group
which in turn is accessed via one of the SIP identities.

6
Audio may be auto-destinated to a single group via VOX or contact closure or to any group(s) via Application Server
events.

7
The Telephone Paging Server detects when a group page has started and begins setting up the IP phones. During
this setup time, the page audio is buffered. Thus, if Valcom speakers and IP phones are both playing the same page
they may be out of sync. To solve this problem the Telephone Paging Server simultaneously sources the audio to
both IP Phones and Valcom Speakers to ensure that they stay synchronized. This is done via the Group Attachment
Editor.
8
Sole access of single channels may be accomplished by creating a group for each channel.

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