add a pause in the dialplan for a second then proceed..
On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield <t...@softins.co.uk> wrote: > In article <20151125133008.6369360.14455.17...@gmail.com>, > Israel Gottlieb <isr...@gmail.com> 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: t...@softins.co.uk - http://www.softins.co.uk > > Play: t...@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: t...@softins.co.uk - http://www.softins.co.uk > Play: t...@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