I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action.
Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway, ringback works fine. >From SIP traces I found that the one that works returns 180 ringing to Asterisk and the one that doesn't work returns 100 trying followed by 183 session progress. It is my understanding that 180 ringing causes ringback to be generated by the callee, while 183 means that the caller has early media and will send ringback through RTP. Anyone have any idea why I wouldn't get ringback in this case? Should Asterisk be passing through the early media to the first caller even though the second caller has not answered? I am not using the "r" option to the Dial command. I have tried it both on and off and get no ringback in either case. I have also tried variations of the progressinband setting. I have listened to the RTP going from the SIP server to Asterisk and I can hear the ringing in it. It seems like Asterisk isn't sending any audio to the first caller until both parties answer. Thanks, Matthew Boedicker _______________________________________________ --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
