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

[Fran Boon]

On Mon, 2004-04-05 at 22:02, Brian Rathman wrote:
> 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.

Sorry, 'T' prevents pass-thru:

http://voip-info.org/wiki-Asterisk+G.729+pass-thru

F