[Asterisk-Users] codecs

[William X Walsh]

On Tue, 2003-02-25 at 09:05, Martin Pycko wrote:
> Ok, if you add
> canreinvite=no
> to your phone entries than it should work.
> Otherwise it's set in default to yes and
> then you make the two SIP devices talk
> in diffrent codecs.

Just tried it.  I had canreinvite=no in each of the sip contexts for
local extensions.  Based on the above, I added it to the sip context for
outgoing FWD calls, and it had no effect on the way the calls failed to
work.  I still get a connection, and there is no audio from either side
of the call unless the remote client is MS Messenger, or asterisk and it
goes straight to voicemail. Calls to FWD users using any other SIP
hardware or software phone still do not work when I'm calling through
asteriskpbx.

Would it help you guys if you saw it in action?  If I gave you the
details of an FWD account, would you take a look?

I've about given up on this after a couple weeks of trying to get it to
interoperate.

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