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

[Andrew Joakimsen]

And if one cannot use a different codec?


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, August 20, 2003 9:51 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] G723 (was SIP using which codec?)

MOH requires that Asterisk transcodes (It also has to transcode to for
PSTN calls and voicemail and playing any sound files).  Asterisk can't
transcode to or from G723.  Nope.  Doesn't work.  May very well never
work.  Use a different codec.

On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote:
> 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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