I have an issue with Dahdi trunk and Asterisk 1.6.0.1 where my analog line is call forwarded on no answer or busy to my sip provider.
When we call in on the analog line, I can see the call begin in the cli, and after 15 seconds I see the call switch over to my sip provider, and after about 30 seconds I get the 3 raising tone signals and the call is hungup. Is that my telco dropping the call for some reason? Incoming calls from the sip provider continue on through its context fine if the call originates through it? I assume the transfer to my sip provider happens as my telco decides it needs to do this. I can investigate that Monday, but why doesn't the incoming sip call continue on through the incoming sip dialplan like it does if I call that number directly and get to voicemail after 45 seconds? Is it possible to make Asterisk answer the incoming dahdi call so the Telco is satisfied but provide ringing to the incoming caller until a handset internally answers or it hits voicemail? Thanks! jlc _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
