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

2015-11-25 Thread Brian :: 
add a pause in the dialplan for a second then proceed..



On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield 
wrote:

> In article <20151125133008.6369360.14455.17...@gmail.com>,
> Israel Gottlieb  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
>
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Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

2015-11-25 Thread Asghar Mohammad 
I had in a same situation and solved by Background 1 sec. silence.

On Wed, Nov 25, 2015 at 5:45 PM, Brian ::  wrote:

> add a pause in the dialplan for a second then proceed..
>
>
>
> On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield 
> wrote:
>
>> In article <20151125133008.6369360.14455.17...@gmail.com>,
>> Israel Gottlieb  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
>
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_
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[asterisk-users] Dialing a call back out on same SIP trunk as it came in

2015-11-25 Thread Tony Mountifield 
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 --
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Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

2015-11-25 Thread Tony Mountifield 
In article <20151125133008.6369360.14455.17...@gmail.com>,
Israel Gottlieb  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

-- 
_
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