[Asterisk-Users] codecs

[William X Walsh]

On Mon, 2003-02-24 at 19:35, William X Walsh wrote:
> On Sat, 2003-02-22 at 17:18, Martin Pycko wrote:
> > Actually now you can use SIP_CODEC variable
> > 
> > eg:
> > 
> > [sip-context]
> > exten => 8500,1,SetVar,SIP_CODEC=alaw
> > exten => 8500,2,VoiceMailMain
> > ....
> > 
> > now when you normally have
> > dissallow=all
> > allow=g729
> > in sip.conf configuration file ... then when you place
> > a call with your SIP phone to 8500 asterisk will
> > force your phone to talk with alaw codec.
> > 
> > regards
> > Martin
> 
> Does this also control the codec asterisk is using to talk to the remote
> device?
> 
> I believe the problem with calling FWD callers outbound from an asterisk
> server is related to that, because I've eliminated everything else I can
> think of.
> 
> Using the above line, it is working at forcing my sjphone to use alaw,
> or ulaw as I configure.
> 
> I noticed when using SJPhone to register directly to FWD, and I call
> another FWD caller is also uses one or the other.
> 
> So I'm wonder if it is possible that asterisk is using another codec
> when trying to talk to the remote device.
> 
> When I use estera softphone with asterisk, and receive a call via fwd, I
> get an error "the media negotiation failed" which seems to support my
> theory.

Ok, let's add to the mystery.

When the remote FWD user is using MSN Messenger's SIP phone, the audio
works fine for calling them outbound using FWD's SIP Service through
asterisk.

So far no other remote SIP enabled soft or hardphone that I have been
able to test it with has worked (except asterisk itself on the remote
side, and then again it seems to depend on the remote client if they
answer.  When the remote client is an asterisk server and the extension
goes to voicemail, no problem.

This is really odd to me  :)

-- 
William Walsh <william at wxw.org>
Jabber: william at wxw.biz