03-20-2009 08:04 AM - edited 03-15-2019 04:59 PM
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
03-20-2009 09:02 AM
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".
05-07-2009 07:48 AM
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:
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:
Call-ID: 653a22804baf356b5bdc2a9e53721a2f@192.168.123.248
CSeq: 801 ACK
Content-Length: 0
05-07-2009 08:10 AM
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
05-10-2009 11:55 PM
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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