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. Thanks, Brian _______________________________________________ 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
