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/[EMAIL PROTECTED],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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
