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[asterisk-users] Queue PJSIP, not all contacts rings


 
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jleed at me.com
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PostPosted: Mon Feb 23, 2015 11:10 am    Post subject: [asterisk-users] Queue PJSIP, not all contacts rings Reply with quote

Hay guys, have question.

When I do regular dial I use $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);

to get all contacts of current endpoint and so I dial to all phones at once,

but if I exec QUEUE, I have just one phone rings, seems like it take first one as Dial app by default, is there way to fix this?
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jcolp at digium.com
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PostPosted: Mon Feb 23, 2015 11:13 am    Post subject: [asterisk-users] Queue PJSIP, not all contacts rings Reply with quote

Nick Awesome wrote:
Quote:
Hay guys, have question.

When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);

to get all contacts of current endpoint and so I dial to all phones
at once,

but if I exec QUEUE, I have just one phone rings, seems like it take
first one as Dial app by default, is there way to fix this?

There is no way to directly do this. The best option is to use a Local
channel into the dialplan which dials instead. Once answered everything
should fall into place.

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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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jleed at me.com
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PostPosted: Mon Feb 23, 2015 12:05 pm    Post subject: [asterisk-users] Queue PJSIP, not all contacts rings Reply with quote

Works, thank you!

Quote:
On Feb 23, 2015, at 7:11 PM, Joshua Colp <jcolp@digium.com> wrote:

Nick Awesome wrote:
Quote:
Hay guys, have question.

When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);

to get all contacts of current endpoint and so I dial to all phones
at once,

but if I exec QUEUE, I have just one phone rings, seems like it take
first one as Dial app by default, is there way to fix this?

There is no way to directly do this. The best option is to use a Local channel into the dialplan which dials instead. Once answered everything should fall into place.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.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


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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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