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SIP GSM Gateway and incoming calls

l.buschi
Level 2
Level 2

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

4 Replies 4

paolo bevilacqua
Hall of Fame
Hall of Fame

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".

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:user_ip@192.168.123.2 SIP/2.0

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

From: "88" <>;tag=1c1cec17

To:

Call-ID: 653a22804baf356b5bdc2a9e53721a2f@192.168.123.248

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: 653a22804baf356b5bdc2a9e53721a2f@192.168.123.248

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: 653a22804baf356b5bdc2a9e53721a2f@192.168.123.248

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:user_ip@192.168.123.2 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: 653a22804baf356b5bdc2a9e53721a2f@192.168.123.248

CSeq: 801 ACK

Content-Length: 0

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

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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