SIP GSM Gateway and incoming calls

Unanswered Question
Mar 20th, 2009
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Hello,

I have a SIP GSM Gateway and I can make outgoing calls without any problem just having configured a dial-peer voip.


I have problems with incoming calls: i make my call, the GSM correctly transfer it to my UC520 and then the call is closed.

with debug voice dialpeer I can see that calling number is "user_ip" and called number is "user_ip" againg so that there are no valid dial peer that can match it.

Can anyone suggest me how to have a translation between "user_ip" and a valid intrnal number?


Thanks


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paolo bevilacqua Fri, 03/20/2009 - 09:02
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Check your GSM GW documentation. You should have the option for setting a destination number for all calls, and you should be able to see the calling number. The best way to check that is "debug ccsip message".

l.buschi Thu, 05/07/2009 - 07:48
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Hi, follwoing is the output of debug ccsip message.


No problem with outgoing call but no success for nicoming call.


UC520-Antea#

May 7 15:47:16.894: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.123.248:5060;rport;branch=z9hG4bK5aa4e201c6

From: "88" ;tag=1c1cec17

To:

Call-ID: [email protected]

Contact:

CSeq: 801 INVITE

Max-Forwards: 70

Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE

Supported: replaces

Content-Type: application/sdp

User-Agent: CM5K (706220)

Content-Length: 397


v=0

o=CMI-SIPUA 46416 0 IN IP4 192.168.123.248

s=SIP CALL

c=IN IP4 192.168.123.248

t=0 0

m=audio 60000 RTP/AVP 0 8 4 18 23 22 2 21 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:23 G726-16/8000

a=rtpmap:22 G726-24/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:21 G726-40/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv


May 7 15:47:16.902: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.123.248:5060;rport;branch=z9hG4bK5aa4e201c6

From: "88" ;tag=1c1cec17

To:

Date: Thu, 07 May 2009 15:47:16 GMT

Call-ID: [email protected]

CSeq: 801 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0



May 7 15:47:16.902: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/UDP 192.168.123.248:5060;rport;branch=z9hG4bK5aa4e201c6

From: "88" ;tag=1c1cec17

To: ;tag=20E0AB54-2632

Date: Thu, 07 May 2009 15:47:16 GMT

Call-ID: [email protected]

CSeq: 801 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=16

Content-Length: 0



May 7 15:47:17.178: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.123.248:5060;rport;branch=z9hG4bK5aa4e201c6

From: "88" ;tag=1c1cec17

To: ;tag=20E0AB54-2632

Call-ID: [email protected]

CSeq: 801 ACK

Content-Length: 0



l.buschi Thu, 05/07/2009 - 08:10
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Ok I solved my problem, now the incoming calls are routed correctly to the IVR, then a can choose an internal number and the call is correclty forwared, but I can hear 1 second of voice and then the call is auto called.




Can anybody help Me?



Thanks


l.buschi Sun, 05/10/2009 - 23:55
User Badges:

Ok I solved my problem, now the incoming calls are routed correctly to the IVR, then a can choose an internal number and the call is correclty forwared, but I can hear 1 second of voice and then the call is auto CLOSED.




Can anybody help Me?



Thanks




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