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Configuring DWG2000C Dinstar

Step-by-step configuration guidelines for DWG2000C Dinstar gsm gateway for SIP server (Trixbox, elastix etc).

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Abdul Dan'O'maru
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3K views3 pages

Configuring DWG2000C Dinstar

Step-by-step configuration guidelines for DWG2000C Dinstar gsm gateway for SIP server (Trixbox, elastix etc).

Uploaded by

Abdul Dan'O'maru
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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Configuring DWG2000C Dinstar Hope it works for you DWG2000C DINSTAR GATEWAY CONFIGURATION - ELASTIX versions used: Asterisk

Server: Elastix 2.0.0-63 IP: 192.168.150.2 Gateway GSM: 2.22.04.02 firmware DWG2000C DinStar IP: 192.168150.8 It is assumed that you have a Asterisk server working properly. Some Fields I left empty here are also empty in the configuration. CONFIGURATION ELASTIX It creates a new SIP trunk pointing GSM Gateway (PBX> Trunks> AddTrunk> Add SIP Trunk) Trunk Description: Celular_DWG_1 Trunk Name : Celular_DWG_1 PEER Details: host = 192.168.150.8 context = from-trunk type = peer insecure = port, invite changes are saved (Submit Changes, Apply Configuration) Creates a new Outbound Route dial using the desired pattern (in Colombia 3XXXXXXXXX ) towards the trunk previously created (PBX> Outbound Routes> Add Route) Route Name: Celular_DWG_1 PIN Set: Cellular Dial Patterns: 3XXXXXXXXX Trunk Sequence: 0 SIP/Celular_DWG_1 changes are saved (Submit Changes, Apply Configuration) It creates a new Inbound Route to route incoming calls, in our case will be routed to the IVR Welcome, will be used to identify the CID 4858999 incoming calls (PBX> Inbound Routes> Add Incoming Route) DID Number: 4858999 Set Destination: IVR: Welcome are saved changes (Submit Changes, Apply Configuration) DWG2000C DINSTAR GATEWAY CONFIGURATION Configure the IP address of the Gateway DinStar (Network Configuration> Local network) Use the following IP address IP Address: 192.168.150.8 Subnet Mask: 255.255.255.0 Default Gateway: 192.168. 150.2 (Note that this is the IP address of Elastix server) IP Trunk is created pointing to our Asterisk server (IP Trunk Configuration> IP Trunk> Add) Index: 31 IP: 192.168.150.2 Port: 5060 Description: COLAST01 Enable KeepAlive:

On mode is configured routing (Routing Configuration> Routing Parameter) IP-> Tel Parameter: Route calls before manipulation Tel-> IP parameter: Route calls before manipulation is configured for routing calls from Elastix to Cell (Routing Configuration> IP- > Call Routing> Add) Index: 0 Description: default Source Prefix: any Source IP: 31 <COLAST01> Destination Prefix: any Port Group: 0 <all> is configured for routing calls from Mobile to Elastix (Routing Configuration> Tel- > IP Routing> Add) Index: 0 Description: default Source Prefix: any Source Port: Any Destination Prefix: any Destination: IP: 31 <COLAST01> permissions are configured for calls from Elastix to Cell (Operation> IP-> Tel Operation) Index: 31 Source Prefix: any Source IP: IP: 31 <COLAST01> Destination Prefix: any Operation: Allow Call Description: COLAST01_Out permissions are configured for calls from cell phones to Elastix (Operation> Tel-> IP Operation) Index: 31 Source Prefix: any Source Port: Port: Any Destination Prefix: any Operation: Allow Call Auto Call: Enable Description: COLAST01_In Configures service settings (System Configuration> Service Parameter) Start Local RTP Port: 8000 Enable silence suppression : Yes Call Progress Tone: USA Preferred Coders: PCMU, PCMA, G729AB, G723.1 Voice Frames per TX: 2 Do Not Answer Incoming Call for GSM Hotline: Yes Enable Incoming GSM Configuration: Yes Auto Routing Outgoing Type: Polling IP to GSM One Stage Dialing: Yes Answer Delay: 5 Call Redirect When All Ports Busy: No Play Voice Prompt for incoming GSM Calls: Yes Parameter DTMF DTMF Method: RFC2833

RFC2833 Payload Type: 101 DTMF Volume: 0dB DTMF interval: 200 NAT Transversal: Disable Other Configuration Enable Private Service: Yes User ID is Phone Number: Yes Only Accept Calls from SIP Server: No Allow Call from GSM to IP without Registration: Yes Allow Call from IP to GSM without Registration: Yes Reject Anonymous call from IP to GSM: No Use # as End Key: Yes No Answer TimeOut: 55 interdigit timeout: 4 Call Delay: 0 are configured SIP account settings (System Configuration> Parameter SIP) SIP Proxy SIP Server Address: 192.168.150.2 SIP Server Port: 5060 Check Status: No Outbound Proxy Address Outbound Proxy: Outbound Proxy Port: 5060 Is Register: No DNS Query type: A query DNS refresh interval: 0 T1: 500 T2: 4000 T4: 5000 TMAX: 32000 Keepalive Interval: 32 Enable 100rel: Not From When Caller ID mode is Available: Tel / Tel mode From Caller ID is Unavailable When: Username Answer Mode: Answered 183 Mode: Immediately Session timer: No Parameters are configured ports (System Configuration> Port Parameter> Detail) All Ports Register same user ID used: Yes SIP User ID: Authenticate ID: Authenticate Password: Local SIP Port: 5060 Tx Gain: +2 dB Rx Gain: +6 dB To VOIP Hotline: 4858999 (This is the number we sent to the inbound route Elastix To PSTN Hotline: Auto-Dial Delay Time. 0 With this we have completed the configuration, restart the Gateway now and are tested for incoming and outgoing calls

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