Louis Yssel Posted March 12, 2009 Report Share Posted March 12, 2009 Since we upgraded PBXnSIP to ver 3, we cannot recieve calls to the Communicator clients anymore. Is there a document available for the integration for ver 3.x as the one available is for ver 2. We did not change anything else, yet incoming calls to Communicator from PBXnSIp no longer works. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 12, 2009 Report Share Posted March 12, 2009 Since we upgraded PBXnSIP to ver 3, we cannot recieve calls to the Communicator clients anymore. Is there a document available for the integration for ver 3.x as the one available is for ver 2. We did not change anything else, yet incoming calls to Communicator from PBXnSIp no longer works. Are you using a country code? That might be a problem; and you can also check on how the PBX represents global numbers on the trunk. OCS likes the "+" notation, at least before R2. A look at the SIP trace should help finding the problem. Quote Link to comment Share on other sites More sharing options...
Louis Yssel Posted March 13, 2009 Author Report Share Posted March 13, 2009 I have aliases in PBXnSIp for all possible formats for internal and external. 0878053505 +878053505 +3505 +8053505 +0878053505 +27878053505 When I run the logging tool in OCS on the Mediation server, the caals from PBXnSIP is not even reaching the OCS server, the call goes directly to Exchange UM. So country code is only used in the last variation of the alias. My OCS URI is +878053505 and the manual registration in PBXnSIP looks like: REGISTER 0878053505 sip:+878053505@172.29.1.200;transport=tcp check-sync When I register a SNOM phone directly to 0878053505, it works 100% Thanks, Quote Link to comment Share on other sites More sharing options...
Louis Yssel Posted March 13, 2009 Author Report Share Posted March 13, 2009 In the PBXnSIP log file I now found the following: [5] 20090313163438: SIP Rx tcp:172.29.1.200:5060: SIP/2.0 100 Trying FROM: "+27829089222"<sip:+27829089222@ysl.yslzone.net:5060;user=phone>;tag=22266 TO: "Louis Yssel"<sip:0878053505@ysl.yslzone.net> CSEQ: 13449 INVITE CALL-ID: 0b085c08@pbx VIA: SIP/2.0/TCP 172.29.1.8:1883;branch=z9hG4bK-7981d3c3feaf5e6c7bb079f42d33ec63;rport CONTENT-LENGTH: 0 [9] 20090313163438: Resolve 781884: aaaa udp 172.28.5.29 2051 [9] 20090313163438: Resolve 781884: a udp 172.28.5.29 2051 [9] 20090313163438: Resolve 781884: udp 172.28.5.29 2051 [9] 20090313163438: Resolve 781885: udp 192.168.0.9 1040 [9] 20090313163438: Resolve 781886: aaaa udp 192.168.0.9 4263 [9] 20090313163438: Resolve 781886: a udp 192.168.0.9 4263 [9] 20090313163438: Resolve 781886: udp 192.168.0.9 4263 [9] 20090313163438: Resolve 781887: aaaa udp 172.27.57.70 2051 [9] 20090313163438: Resolve 781887: a udp 172.27.57.70 2051 [9] 20090313163438: Resolve 781887: udp 172.27.57.70 2051 [9] 20090313163438: Resolve 781888: aaaa udp 172.27.58.59 2051 [9] 20090313163438: Resolve 781888: a udp 172.27.58.59 2051 [9] 20090313163438: Resolve 781888: udp 172.27.58.59 2051 [9] 20090313163438: Resolve 781889: aaaa udp 172.27.57.71 2051 [9] 20090313163438: Resolve 781889: a udp 172.27.57.71 2051 [9] 20090313163438: Resolve 781889: udp 172.27.57.71 2051 [5] 20090313163438: SIP Rx tcp:172.29.1.200:5060: SIP/2.0 488 Invalid incoming Gateway SDP: Did not find common codecs in media stream line in SDP offer FROM: "+27829089222"<sip:+27829089222@ysl.yslzone.net:5060;user=phone>;tag=22266 TO: "Louis Yssel"<sip:0878053505@ysl.yslzone.net>;tag=a6d092cddb CSEQ: 13449 INVITE CALL-ID: 0b085c08@pbx VIA: SIP/2.0/TCP 172.29.1.8:1883;branch=z9hG4bK-7981d3c3feaf5e6c7bb079f42d33ec63;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 20090313163438: Call 0b085c08@pbx#22266: Clear last INVITE [9] 20090313163438: Resolve 781890: url sip:+878053505@172.29.1.200;transport=tcp [9] 20090313163438: Resolve 781890: a tcp 172.29.1.200 5060 [9] 20090313163438: Resolve 781890: tcp 172.29.1.200 5060 [5] 20090313163438: SIP Tx tcp:172.29.1.200:5060: ACK sip:+878053505@172.29.1.200;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.29.1.8:1883;branch=z9hG4bK-7981d3c3feaf5e6c7bb079f42d33ec63;rport From: "+27829089222" <sip:+27829089222@ysl.yslzone.net:5060;user=phone>;tag=22266 To: "Louis Yssel" <sip:0878053505@ysl.yslzone.net>;tag=a6d092cddb Call-ID: 0b085c08@pbx CSeq: 13449 ACK Max-Forwards: 70 Contact: <sip:0878053505@172.29.1.8:1883;transport=tcp> Content-Length: 0 [5] 20090313163438: INVITE Response 488 Invalid incoming Gateway SDP: Did not find common codecs in media stream line in SDP offer: Terminate 0b085c08@pbx [7] 20090313163438: Other Ports: 35 What would the Invalid incoming gateway SDP be. I have not set any preferred codec in the trunk settings. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 13, 2009 Report Share Posted March 13, 2009 What would the Invalid incoming gateway SDP be. I have not set any preferred codec in the trunk settings. The SDP offer is in the INVITE packet. It should be just "above" the trace you showed. Quote Link to comment Share on other sites More sharing options...
Louis Yssel Posted March 16, 2009 Author Report Share Posted March 16, 2009 It seems like the codec that PBXnSIp uses to talk to OCS is not right. In the OCS trunk I do not specifiy and Codec Preference, and at System level I now changed the preferred codec to G.711. but still not. m=audio 59148 RTP/AVP 18 101 I find the above line all over in the SIP trace and I know it is then trying codec 18 (G.729) which OCS does not seem to like, and then 101, which I do not know what it is. Thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 16, 2009 Report Share Posted March 16, 2009 It seems like the codec that PBXnSIp uses to talk to OCS is not right. In the OCS trunk I do not specifiy and Codec Preference, and at System level I now changed the preferred codec to G.711. but still not. m=audio 59148 RTP/AVP 18 101 I find the above line all over in the SIP trace and I know it is then trying codec 18 (G.729) which OCS does not seem to like, and then 101, which I do not know what it is. Thanks Yes, that means that if offers only G.729 and OCS does not accept that. You must also offer other codecs like G.711. Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted April 7, 2009 Report Share Posted April 7, 2009 Hello Louis, are you still facing this problem? Or did you found a solution. I am not sure, but I guess I have seen something similiar. If possible, please update us with your status! Best regards, Jan Quote Link to comment Share on other sites More sharing options...
safiman Posted April 27, 2009 Report Share Posted April 27, 2009 Hello, i'am in intership in the compagny, i deployed ocs 2007 R2, i integrated ocs with mitel 3300, all calls works : pc to pc / ip-phone to pc, except outboun call, from communicator to iP-Phone (pc to ip-Phone), i don't have a media gateway just a mediation server connected directly to Mitel with sip trunking. so i spend a month (April) but i didn't find any solution, can you help me please ? Quote Link to comment Share on other sites More sharing options...
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