[Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

[Matt Riddell [NZ]]

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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