[Freeswitch-users] Where is that codec list coming from?
Kristian Kielhofner
kristian.kielhofner at gmail.com
Wed Dec 16 11:41:00 PST 2009
Sure... The call comes up as PCMU:
INVITE sip:5888 at 10.70.0.99 SIP/2.0
Call-ID: 80ea31a017f6de1d53e4a9c52f00
CSeq: 1 INVITE
From: sip:9413122830 at 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:5888 at 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:9413122830 at 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 at 10.70.0.99>;avaya-cm-alert-type=external
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: sip:9413122830 at smh.sip.local
P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52"
History-Info: <sip:5888 at 10.70.0.99>;index=1,"5888"
<sip:5888 at 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.minessale at 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
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