I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical machine with a similar configuration. Thoughts? I know I posted this yesterday but was hoping for some more creative comments!
Zip*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip.callwithus.com:5060 N xxxx 105 Registered Tue, 07 Dec 2010 02:36:43 1 SIP registrations. my sip.conf [general] context=default allowoverlap=no ;bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a NAT ro ;register => xxxx:[email protected] register => xxxx:[email protected] [callwithus] type=friend host=sip.callwithus.com username=xxxx secret=31 qualify=no insecure=invite my extensions.conf [general] [globals] CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] exten => s,1,Answer() exten => s,n,Dial(SIP/callwithus/12222222222) exten => s,n,Wait(2) exten => s,n,Hangup() -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
