In article <[email protected]>, Israel Gottlieb <[email protected]> wrote: > Try putting progress instead of answer
Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to another > number via the same SIP trunk as it came in on. e.g. > > [from-siptrunk] > exten => 0123456789,1,NoOp > exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) > > Now, if I use a different SIP trunk for the outbound call, than the > inbound call came on, the call is set up fine - the Answer signal from the > called party gets propagated back to the caller, and they can hear each > other. > > But if the outbound SIP trunk is the same as the one the call came in on, > the caller doesn't hear any progress, and has no notification of when the > call was answered. Neither can the parties hear each other. > > I have tried this on two different machines using two different SIP > providers. > > However, if I change the above NoOp to be Answer(100), i.e. answer the > inbound call before placing the outbound Dial, the caller hears progress > and when the called party answers, they hear each other fine. > > Of course, if the called party is busy, the caller just hears in-band > busy tone, as the caller's inbound call was already answered. > > Can anyone explain why I need the Answer? It feels wrong that I should. > > The siptrunk entry contains canreinvite=no and directmedia=no. > > The version of Asterisk on these boxes is 10.5.1, if that's relevant. > > Thanks for any insight! > > Cheers > Tony > > -- > Tony Mountifield > Work: [email protected] - http://www.softins.co.uk > Play: [email protected] - http://tony.mountifield.org > > -- > _____________________________________________________________________ > -- 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 > -- > _____________________________________________________________________ > -- 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 > -- Tony Mountifield Work: [email protected] - http://www.softins.co.uk Play: [email protected] - http://tony.mountifield.org -- _____________________________________________________________________ -- 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
