[asterisk-users] Dialing a call back out on same SIP trunk as it came in

[Brian ::]

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
> >
> > --
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>
> --
> 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
>
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