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 Post subject: Unable to find a codec translation path g729 to GSM
PostPosted: Mon Apr 08, 2013 4:45 pm 
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Joined: Mon Apr 08, 2013 4:38 pm
Posts: 6
Greetings,

I am attempting to demo the A2Billing IVR and have set up an IPkall.com number that is routing to my A2Billing server. When the call gets to the server I receive this error:

== Using SIP RTP CoS mark 5
-- Executing [0470523646@a2billing:1] AGI("SIP/ipkall-0000025e", "a2billing.php,1") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/a2billing.php
[Apr 8 12:48:02] WARNING[17087]: channel.c:5107 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Apr 8 12:48:02] WARNING[17087]: file.c:958 ast_streamfile: Unable to open prepaid-enter-pin-number (format 0x100 (g729)): No such file or directory
[Apr 8 12:48:02] WARNING[17087]: channel.c:5107 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Apr 8 12:48:02] WARNING[17087]: file.c:958 ast_streamfile: Unable to open prepaid-enter-pin-number (format 0x100 (g729)): No such file or directory
[Apr 8 12:48:02] WARNING[17087]: channel.c:5107 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Apr 8 12:48:02] WARNING[17087]: file.c:958 ast_streamfile: Unable to open prepaid-enter-pin-number (format 0x100 (g729)): No such file or directory
-- <SIP/ipkall-0000025e>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing, 0470523646, 1) exited non-zero on 'SIP/ipkall-0000025e'

It appears the g729 codec needs to be installed and that the call needs the GSM formatted audio files to transcode. Does this require a g729 license to be installed? If it is required, can I force a different codec from the originator to avoid license cost?


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 Post subject: Re: Unable to find a codec translation path g729 to GSM
PostPosted: Mon Apr 08, 2013 6:34 pm 
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Joined: Mon Mar 02, 2009 8:56 pm
Posts: 227
Asterisk does not come with G729 because of license restrictions. There is an opensource version here - http://asterisk.hosting.lv/ - but you should confirm you comply with any license restrictions. You can also buy licenses from Digium.

You can force a codec on a trunk by using (ulaw for example) -

disallow=all
allow=ulaw

That obvisouly requires that the remote PBX supports ulaw (g711) also.


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 Post subject: Re: Unable to find a codec translation path g729 to GSM
PostPosted: Wed Apr 10, 2013 6:29 pm 
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Joined: Mon Apr 08, 2013 4:38 pm
Posts: 6
The answer to my original question is "Yes, you need a Digium license installed to do the transcoding. Additionally, if you open up different codecs on the trunk you can avoid this".

Bucasia, I was able to bypass this issue by doing as you instructed. When allow=ulaw,alaw,gsm is in the sip.conf with the trunk the call does reach the switch and plays the proper IVR files according to the order in a2billing.php. I've hit an entirely different issue now with dead-air from the IVR. The logs show the audio files playing, but the person dialing in gets dead air. That issue has been logged here if you want to take a look:

viewtopic.php?f=33&t=10296

Thanks for your help!


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 Post subject: Re: Unable to find a codec translation path g729 to GSM
PostPosted: Wed Apr 10, 2013 6:43 pm 
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Joined: Mon Mar 02, 2009 8:56 pm
Posts: 227
Is there any NAT involved? Is your Asterisk server behind a NAT router?

This is a common cause of one-way or no audio.


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