SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14
My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my
dial plan set up so that when outside callers dial the DiD, the call is
answered by my auto-attendant. The caller can then select who they'd like
to speak to a
If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.
I have had great success getting this to work using IA
I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP
re-invites.
I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu
recording and transfers the call to the external line the caller selects.
Since both sides of the call are external, I want to use re-i
