[Asterisk-Users] Forcing a codec

[Chris Stenton]

I think if you do 

disallow=all
allow=ulaw
allow=ilbc

in the "general" section on sip0 latter versions of * seem to take
notice of the order in the general section.

Chris

On Fri, 2004-10-01 at 20:56, Eric wrote:
> Hi,
> 
> I'm having trouble explicitly forcing a codec between sip devices.  Am
> I missing something or is this not really possible?
> 
> I have a grandstream registering to asterisk, named sip0.  Sip0 registers,
> via sip, to another asterisk box, sip1.  When I place a call from the
> grandstream, it will travel through sip0 to sip1, where it is then placed
> to the PSTN.  Nothing can reinvite, this path is forced for the entire
> call.
> 
>  +--------+         +--------+         +--------+
>  |   gs   | <-----> |  sip0  | <-----> |  sip1  |
>  +--------+         +--------+         +--------+
> 
> I would like the RDP traffic between the gs and sip0 to be encoded using
> ILBC and the traffic between sip0 and sip1 to use G.711.  I can force the
> gs/sip0 path to ILBC be allowing only that codec in the gs's sip config,
> however, even when I specify in the sip config that sip1 can only use ulaw,
> it uses ILBC, as observed from a `sip show peers`.  sip1 allows both ILBC
> and ULAW.
> 
> Is there any way to force sip0 to reencode the audio stream?
> 
> sip0 is running asterisk 1.0.1 and the gs is the latest 1.0.5.11 code.
> sip1 is running an older CVS version.
> 
> Anyone have an ideas?
> 
> - Eric
> 
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