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

[Matthew J. Roth]

Kamlesh,

Please provide SIP traces of both call legs for a failed call.

Your last message only included a SIP trace of the call leg from the SIP
softphone to the Asterisk server.  There was no SIP trace for the call leg from
the Asterisk server to the ITSP and, as shown below, that is probably where the
answer to your problem can be found.

First, the call leg from the SIP softphone to the Asterisk server successfully
negotiated G.729 as the codec:

> [May 28 11:51:34] Found RTP audio format 18
> ...
> [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)

However, the "call.php" AGI script then failed to create the call leg from the
Asterisk server to the ITSP:

> [May 28 11:51:34]     -- Executing AGI("SIP/100-0000115f", "call.php")
> [May 28 11:51:34]     -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php
> [May 28 11:51:34]     -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)
> [May 28 11:51:34]   == Using SIP RTP CoS mark 5
> [May 28 11:51:34]     -- Couldn't call yyy.yyy.yyy.yyy/12127773456
> [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071ad6 at xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
> [May 28 11:51:34]   == Everyone is busy/congested at this time (0:0/0/0)
> [May 28 11:51:34]     -- <SIP/100-0000115f>AGI Script call.php completed, returning 0
> [May 28 11:51:34]     -- Auto fallthrough, channel 'SIP/100-0000115f' status is 'CHANUNAVAIL'

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer