[Asterisk-Users] G723 (was SIP using which codec?)

[Andrew Joakimsen]

Actually I got it working right before I gave up (I had the wrong line
in my config commented out)

But now I get these messages when I try to playback a recording:

NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable
to find a path from GSM to G723
WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open
transfer (format G723): No such file or directory
WARNING[16401]: File app_playback.c, Line 83 (playback_exec):
ast_streamfile failed on SIP/packet8.net-dab9 for transfer

And when I try to play music on hold:

NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable
to find a path from SLINR to G723
WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable to
set 'SIP/packet8.net-dab9' to signed linear format


This is the missing link in my system, I greatly appreciate any help
that can be provided.

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Todd
Sent: Wednesday, August 20, 2003 4:11 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP using which codec?

At 7:27 AM +0000 8/20/03, WipeOut . wrote:
>
>>  Is there a way to determine what codec the remote server wants to
use in
>>  a SIP session for an incoming call by looking at something, possiby
sip
>>  debug?
>>
>
>Take a look in the archives this was covered a couple of days ago..
>
>the command you are looking for is "sip show channels".. and then 
>look in the format column.. the formula for determining the format 
>was posted in the previous discussion and i can't rememebr it off 
>the top of my head..
>
>Later..
>

I think that the question is a bit more subtle than that.  The 
question says "wants to use", not "does use."

Currently, I think the only way you'll find this is with a SIP debug, 
looking at the SDP request.

JT
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