HI folks! my topology is:
softswitch (BROADSOFT) -- [sip trunk] -- Asterisk
I need to connect phone calls using g729 codec. Debugging some calls we
found that calls cant connect because of codec incompatibility. Our Sip
provider send us annexb=yes when a call is comming and our asterisk send
annexb=no. Im running asterisk 1.4.21.1. Output debug shows:
To: <sip:[EMAIL PROTECTED];user=phone>
CSeq: 1 INVITE
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R006
Max-Forwards: 69
Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,ME
SSAGE,REFER
Content-Length: 274
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 194579 194579 IN IP4 189.8.113.170
s=Sip Call
c=IN IP4 XXX.X.XXX.170
t=0 0
m=audio 49256 RTP/AVP 18 8 0 97
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
Via: SIP/2.0/UDP
XXX.X.XXX.170:5060;branch=z9hG4bKo2echm202o5g2dc38701.1;received=XXX.X.XXX.1
70
From: <sip:000@ XXX.X.XXX.170;user=phone>;tag=25f94692
To: <sip:7002@ XXX.X.XXX.177;user=phone>;tag=as0de67360
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Supported: replaces
Contact: <sip:7002@ XXX.X.XXX.177>
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 10183 10183 IN IP4 XXX.X.XXX.177
s=session
c=IN IP4 XXX.X.XXX.177
t=0 0
m=audio 10772 RTP/AVP 18 97
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Thanks for any help!
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
[EMAIL PROTECTED]
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
