[Asterisk-Users] Extensions.conf sending calls to Cisco AS5300

[Brian Rathman]

I have my server configured to send to send all PSTN traffic to my Cisco
AS5300 gateway via SIP. I use the following line in the extensions.conf file
to accomplish this:

exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@10.1.1.1,240,T)

Unfortunately, when I removed the T from the end of the statement, the calls
still complete, but they drop as soon as the called party answers the phone.
I thought that the T had something to do with a timeout, but I have also
seen documentation referencing that it allows * to stay in the middle of the
call to determine if the customer use the # key, etc. I have not been able
to find the detailed documentation that I was looking for on this subject.
Can someone please direct me to this?

Also it is my understanding, that if * stays in the middle of the call, I
can not use the g729 codec without licensing from Digium. If this is the
case, is there a way that I can use g729 in pass thru and still complete
calls to the gateway? Any help would be greatly appreciated.

Thanks,
Brian