Sure... The call comes up as PCMU: INVITE sip:[email protected] SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:[email protected];tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp> To: "5888" <sip:[email protected]> Via: SIP/2.0/UDP 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 Content-Length: 206 Content-Type: application/sdp Contact: <sip:[email protected];transport=tcp> Max-Forwards: 70 User-Agent: Avaya CM/R015x.02.0.947.3 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: timer,replaces,join,histinfo,100rel Alert-Info: <cid:[email protected]>;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:[email protected] P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: <sip:[email protected]>;index=1,"5888" <sip:[email protected]>;index=1.1
v=0 o=- 1 1 IN IP4 10.70.0.69 s=- c=IN IP4 10.70.0.22 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 We don't support G729 so this call comes up as PCMU when we answer and then that codec is first in the codec list... On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale <[email protected]> wrote: > can you do another trace to show the inbound invite too? > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
