[asterisk-users] G.729 codec in pass-thru mode

[Eric Wieling]

Show us the sip debug for a failed call.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kamlesh Kumar
Sent: Monday, May 27, 2013 2:20 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] G.729 codec in pass-thru mode

Hello,
Trying to use g729 in pass-thru mode.
Call flow:
SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI.
== Using SIP RTP CoS mark 5
-- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call.php
-- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456)
-- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0)
-- <SIP/100-00000000>AGI Script call.php completed, returning 0
-- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL'
 
If I use, ulaw, call works fine.
 
Thanks,
Kamlesh