Hi,
We have an audio codec problem in our calls involving our telephony system and our providers Cisco AS5300 trunk.
Our telephony system is Asterisk version 1.6.2.14 . G729a Codec is licensed, bought from Digium.
AS5300 trunk is our outgoing calls sip peer.
We set the sip peer to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted.
We can hear audio only when we both allow ulaw codec.
We do not have this problem on our other providers using asterisk and other non-cisco systems.
Anyone else having this same problem?
Here's our config (both sides) and the sip debug output:
Asterisk sip.conf setting:
type=peer
disallow=all
allow=g729
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
nat=no
canreinvite=yes
context=from-trunk-sip-iaccess
AS5300 Settings:
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g723r53
codec preference 4 g723r63
codec preference 5 g723ar53
codec preference 6 g723ar63
SIP Debug Output:
v=0
o=root 1686167830 1686167830 IN IP4 yyy.yy.yy.yy
s=Asterisk PBX 1.6.2.14
c=IN IP4 yyy.yy.yy.yy
t=0 0
m=audio 14212 RTP/AVP 18 0 8 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
v=0
o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx
s=SIP Call
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18132 RTP/AVP 18
c=IN IP4 xxx.xxx.xxx.xxx
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
