[Asterisk-Users] SIP reinvite still does not occour

[Patrick]

On Fri, 2006-06-30 at 10:39 +0200, Roger Schreiter wrote:
> Hoa Thai Duy schrieb:
> > Roger
> > 
> > If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
> > issue no re-INVITE, for sure.
> > 
> > Pls. change 
> > 
> > Disallow=all
> > Allow=gsm (only one codec)
> 
> 
> Hi,
> 
> yes, to avoid transcoding problems I only have one
> codec, just alaw. Anything else is disallowed.
> That's why I don't understand, why there is no reinvite.
> 
> Thanks for answering!

Iirc if you have something like a "t" or "T" in your Dial command in
extensions.conf than canreinvite will not work because Asterisk has to
stay in the middle to take care of the "t" or "T". Remove these (and
maybe othger) options from the Dial command and give it a try again.

Regards,
Patrick