I had in a same situation and solved by Background 1 sec. silence. On Wed, Nov 25, 2015 at 5:45 PM, Brian :: <[email protected]> wrote:
> add a pause in the dialplan for a second then proceed.. > > > > On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield <[email protected]> > wrote: > >> 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 >> > > > -- > _____________________________________________________________________ > -- 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 >
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