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

Asghar Mohammad asghar144 at gmail.com
Wed Nov 25 11:42:27 CST 2015


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
>> >
>> > --
>> > _____________________________________________________________________
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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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>
>
> --
> _____________________________________________________________________
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
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