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
Brian ::
2015-Nov-25 16:45 UTC
[asterisk-users] Dialing a call back out on same SIP trunk as it came in
add a pause in the dialplan for a second then proceed.. On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield <tony at softins.co.uk> wrote:> In article <20151125133008.6369360.14455.17239 at gmail.com>, > Israel Gottlieb <isrlgb at 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: 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 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151125/d8c8430e/attachment.html>
Asghar Mohammad
2015-Nov-25 17:42 UTC
[asterisk-users] Dialing a call back out on same SIP trunk as it came in
I had in a same situation and solved by Background 1 sec. silence. On Wed, Nov 25, 2015 at 5:45 PM, Brian :: <bc at iptel.co> wrote:> add a pause in the dialplan for a second then proceed.. > > > > On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield <tony at softins.co.uk> > wrote: > >> In article <20151125133008.6369360.14455.17239 at gmail.com>, >> Israel Gottlieb <isrlgb at 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: 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 >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151125/f4ffc8b1/attachment.html>