[Asterisk-Users] One-way audio with SIP client only on certaincalls

[Nabeel Jafferali]

> Try this:
> 
> canreinvite=no

As I mentioned in my initial email, I tried that, and adding that line
eliminated 1 of 2 problems. The other problem, that of one-way audio
when a call is carried into the server from and IAX gateway provider to
that SIP client, will not go away.

-- 
Nabeel Jafferali
tel: 647.722.8457 x201
     718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net