On Friday 23 October 2009 14:19:45 Ross Beer wrote: > Hi, > > Here is the sip trace for the calls, it strange as all other phone work > except X-Lite, it appears that sound is not getting from the softphone to > the server. Though if the softphone talks directly to asterisk it works ok. > This is the case if MediaProxy is used or not. > > There is a lack of codecs in the invite which is strange as I have 4 > enabled on the server and the softphone.
Seems you don't look on the wright place .. because your X-Lite is offering PCMU ... just look at the m= line on the first invite. > INVITE sip:160@<SERVER IP ADDRESS> SIP/2.0 > Via: SIP/2.0/UDP <ROUTER IP > ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport > Max-Forwards: 69 > Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302> > To: "160"<sip:160@<SERVER IP ADDRESS>> > From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;tag=ad038800 > Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM. > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO Content-Type: application/sdp > User-Agent: X-Lite release 1103k stamp 53621 > Content-Length: 236 > v=0 > o=- 0 2 IN IP4 192.168.1.222 > s=CounterPath X-Lite 3.0 > c=IN IP4 <ROUTER IP ADDRESS> > t=0 0 > m=audio 10006 RTP/AVP 0 101 > a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv Umm ... umm ... by ROUTER IP ADDRESS do you mean your public IP on your NAT router ? ... does your router have SIP-ALG activated ?, that could explain the problem. This INVITE is the one that just arrive at your proxy? ... if so .. your NAT router is doing sip-alg .. so mangling all the SIP dialog, and they usually does a very bad job with that. -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users