Tony Mountifield
2015-Nov-25 13:14 UTC
[asterisk-users] Dialing a call back out on same SIP trunk as it came in
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: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org
Israel Gottlieb
2015-Nov-25 13:30 UTC
[asterisk-users] תשובה: Dialing a call back out on same SIP trunk as it came in
Try putting progress instead of answer ? ????? ?????? ? ???: Tony Mountifield ????: ??? ?????, 25 ??????? 2015 08:14 ??: asterisk-users at lists.digium.com ??? ?: Asterisk Users Mailing List - Non-Commercial Discussion ????: [asterisk-users] Dialing a call back out on same SIP trunk as it came in 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: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - 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
Tony Mountifield
2015-Nov-25 14:27 UTC
[asterisk-users] Dialing a call back out on same SIP trunk as it came in
In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote:> Try putting progress instead of answerYes, 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: tony at softins.co.uk - http://www.softins.co.uk > Play: tony at mountifield.org - 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: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org