asterisk users - Mar 2006 - Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

So, when I get no comments on this at all, either here or on any of the forums,
does that mean nobody knows what I'm talking about??  Or does nobody know
the answer??  Or is it just a stupid question and nobody wants to bother telling
me where to look??

It *is* a question that I have to answer somehow; I've read all through TFOT
and see nothing relevant to this issue.  It's silly to spend $15000 on a
G723 license just so I can play back menu messages from Asterisk (since the
actual call decoding is done by the external boxes, which have already paid the
licensing fees).

Dan Miller
  ----- Original Message ----- 
  From: Dan Miller 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, March 06, 2006 10:11
  Subject: PLEASE respond: how to get Asterisk to change coders on RTP handoff??


  I have a hardware FXO/FXS which handle my voip calls, and they support G723
internally.  Asterisk hands off these calls just fine, and everything works, as
long as I don't want PBX menues available...  The problem is, once I want it
to return messages, it will only return them as GSM... which is fine, since my
FXO/FXS support multiple coders.  However, even though Asterisk lets me specify
a list of valid coders, it will only use one...

  I want Ast to use GSM to playback messages, then when it hands off the call to
the endpoints, it should tell them to use G723 in the RE-INVITE messages... I
don't see any way to get it to do this; *is* there some way??
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Dan Miller wrote:
> So, when I get no comments on this at all, either here or on any of the
forums, does that mean nobody knows what I'm talking about??  Or does nobody
know the answer??  Or is it just a stupid question and nobody wants to bother
telling me where to look??
> 
> It *is* a question that I have to answer somehow; I've read all through
TFOT and see nothing relevant to this issue.  It's silly to spend $15000 on
a G723 license just so I can play back menu messages from Asterisk (since the
actual call decoding is done by the external boxes, which have already paid the
licensing fees).
You can not really currently change codecs mid call (in most situations)
although work has been progressing in this area for some time.

Theoretically you should be able as others have based IAX devices around
this concept, but I don't think its available for sip.

Your other option would be to convert the audio files from GSM to G723.1
and that way, playing them would not require transcoding.

-- 
Cheers,

Matt Riddell
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