> I am trying to route incoming DID call (on a analog channel) through > Asterisk to an outside (analog) line. My extensions.conf is something like > the following: > > exten => 500,1,Dial,Zap/g1/3105551010 > > In this case the incoming DID call extension is 500. I am able to dial out > and connect with the incoming call, however, the voice conversation is only > one way. The called party is not able to hear the calling party. > > Does anyone have any suggestions of how to better route the incoming DID > calls to external POTS lines. Basically what I am trying to do is to forward > DID calls to specific external PSTN phone numbers. A one to one forwarding > scheme.
It's almost impossible to provide any answers as you haven't provided enough config data for anyone to technically understand what you are currently doing or suggest alternative choices. Include something that would show us where the DID's are coming from (itsp, telco) and the associated *.conf entries, the actual extensions.conf context entries used for those incoming DID calls, *.conf entries for the external pots lines, etc. You might also give us a clue what country the * box is located. Some countries supply disconnect supervision on pots lines while others don't. (There is at least some possibility what you're trying to accomplish won't work correctly regardless of how you configure * due to the lack of disconnect supervision on pots lines, etc.) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
