[asterisk-users] DIALSTATUS on CANCEL

[Bryant Zimmerman]

Vardan

I have not use AEL so it is a bit hard to follow with the formatting the 
way it is but it looks like correct.
Please note the "h" extension only appears to run if a call is connected so 
I do not know when the "CANCEL" would ever be set. 
There may be someone else who can speak to this. It also appears thet 
${DIALSTATUS} may not be set if the call is not allowed to time out or 
dialed. To me it would make sense to set the inital state of the 
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but I 
may be missing the point on this can anyone else speak to it?

Bryant

----------------------------------------
 From: "Vardan Harutyunyan" <hvardan71 at gmail.com>
Sent: Thursday, December 23, 2010 2:11 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL

I have make test in AEL.

context fu {

_000./userN => {
Dial(SIP/${EXTEN:3}@Prov);
Noop(${DIALSTATUS});
};
h => {
Noop(${DIALSTATUS});
};
};

And look CLI
-- Executing [00018185402020 at fu:1] NoOp("SIP/userN-b6317738", "") 
in new stack
-- Executing [00018185402020 at fo:2] Dial("SIP/user3-b6317738", 
"SIP/18185402020 at Prov") in new stack
-- Called 18185402020 at Prov
-- SIP/Prov-082a83b8 is making progress passing it to 
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on 
'SIP/user3-b6317738'
-- Executing [h at fu:1] NoOp("SIP/userN-b6317738", "CANCEL") in new stack

I think, I am right

-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: info at eif.am
www.eif-it.com

Bryant Zimmerman wrote:
> The Dial Status is not set when accessing it from the h extension.
>
> Bryant
>
> ------------------------------------------------------------------------
> *From*: "Vardan Harutyunyan" <hvardan71 at gmail.com>
> *Sent*: Wednesday, December 22, 2010 10:39 AM
> *To*: asterisk-users at lists.digium.com
> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
>
> Try to use h extension
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: info at eif.am
> www.eif-it.com
>
> Michael wrote:
>> Hi Nikhil,
>>
>> Both debug and verbose are set to 20. That's all I got, but as you can
>> see, for the other types of reasons, the DIALSTATUS got a value (and we
>> see the events). I'm pretty sure it's a bug.
>>
>> Michael
>>
>> On Wed, Dec 22, 2010 at 9:01 AM, Nikhil <d.nikhil at cem-solutions.net
>> <mailto:d.nikhil at cem-solutions.net>> wrote:
>>
>> Hi
>> Enable debug level to more than 1 ,you may get something.
>>
>> Thanks
>> Nikhil
>>
>> On 12/22/2010 11:26 AM, Michael wrote:
>>
>> Spawn extension (incoming-private, 11111111, 3) exited non-zero
>> on 'SIP/Proxy-00000031'
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
>
>
> --
> _____________________________________________________________________
> -- 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/20101223/b5895269/attachment.htm>