[Asterisk-Users] Why Won't Asterisk REINVITE?

[Eric "ManxPower" Wieling]

T/t/H/h and other options to Dial require Asterisk to stay in the RTP 
stream.

George Pajari wrote:
> We are trying to use Asterisk to set up a call between two SIP devices 
> and then step out of the path.
> 
> - all systems have public IP addresses (no firewalls, no NAT).
> - sip.conf has "canreinvite=yes" for both devices
> - ulaw is the only permitted codec so we do not have transcoding issues 
> (and a "sip show channels" confirms both legs at ulaw)
> 
> yet a SIP trace shows that Asterisk does even try to issue a reinvite.
> 
> What else should we look at to see where things are going wrong?
>