Sure... The call comes up as PCMU: INVITE sip:5...@10.70.0.99 SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:9413122...@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp> To: "5888" <sip:5...@10.70.0.99> 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:9413122...@10.70.0.69;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:external@10.70.0.99>;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:9413122...@smh.sip.local P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: <sip:5...@10.70.0.99>;index=1,"5888" <sip:5...@10.70.0.99>;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 <anthony.miness...@gmail.com> 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
