yah so the codec chosen by the inbound leg is always offered in the outbound
sdp to try and prevent transcoding.
if you set {absolute_codec_string=G722} in the bridge string you will bypass
this feature.On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner < [email protected]> wrote: > Sure... The call comes up as PCMU: > > INVITE sip:[email protected] <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] <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] <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] <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] <sip%[email protected]> > >;index=1,"5888" > <sip:[email protected] <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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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