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?
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