[Asterisk-Users] Forcing a codec (take 2)

[Eric]

I'm reposting this to the list..  My spam filters didn't like the list host. :(

If anyone was able to respond to the mail below, can you send it again
please?

Thanks.


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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