Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten => s-CANCEL,1, NoOp()
exten => s-CANCEL,n, Return()
exten => s-NOANSWER,1, NoOp()
exten => s-NOANSWER,n, Return()
exten => s-BUSY,1, NoOp()
exten => s-BUSY,n, Return()
This is what we get on a BUSY call:
-----------------------------------
-- Executing [11111111 at incoming-private:3]
Dial("SIP/Proxy-0000002b",
"SIP/1001,50") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- Got SIP response 486 "Busy Here" back from 10.0.0.1
-- SIP/1001-0000002c is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [11111111 at incoming-private:4]
NoOp("SIP/Proxy-0000002b",
"BUSY") in new stack
-- Executing [11111111 at incoming-private:5]
Gosub("SIP/Proxy-0000002b",
"incoming-status,s-BUSY,1") in new stack
This is what we get on a NO ANSWER call:
---------------------------------------
-- Executing [11111111 at incoming-private:3]
Dial("SIP/Proxy-0000002f",
"SIP/1001,30") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-00000030 is ringing
-- Nobody picked up in 30000 ms
-- Executing [11111111 at incoming-private:4]
NoOp("SIP/Proxy-0000002f",
"NOANSWER") in new stack
-- Executing [11111111 at incoming-private:5]
Gosub("SIP/Proxy-0000002f",
"incoming-status,s-NOANSWER,1") in new stack
This is what we get on a CANCEL call:
-------------------------------------
-- Executing [11111111 at incoming-private:3]
Dial("SIP/Proxy-00000031",
"SIP/1001,30") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- SIP/1001-00000032 is ringing
== Spawn extension (incoming-private, 11111111, 3) exited non-zero on
'SIP/Proxy-00000031'
There's no event indicating that a DIALSTATUS is generated and the call
simply doesn't go to the next step in the dialplan. Unless I'm missing
something, it seems to me that it might be a bug.
I would be happy to get feedback from other users of the DIALSTATUS value
(or Digium), especially in the CANCEL scenario.
Thank you,
Michael
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Anyone?? Thanks. On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question <voip.question at gmail.com>wrote:> Hello, > > We have a strange situation (asterisk 1.6.2.14), where we get a result for > DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. > > This is the (relevant) test dialplan: > -------------------------------- > [incoming-private] > exten => _X., n, Dial(SIP/1001,30) > exten => _X., n, NoOp(${DIALSTATUS}) > exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) > > [incoming-status] > exten => s-CANCEL,1, NoOp() > exten => s-CANCEL,n, Return() > exten => s-NOANSWER,1, NoOp() > exten => s-NOANSWER,n, Return() > exten => s-BUSY,1, NoOp() > exten => s-BUSY,n, Return() > > > This is what we get on a BUSY call: > ----------------------------------- > -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-0000002b", > "SIP/1001,50") in new stack > == Using SIP RTP CoS mark 5 > == Using SIP VRTP CoS mark 6 > == Using UDPTL CoS mark 5 > -- Called 1001 > -- Got SIP response 486 "Busy Here" back from 10.0.0.1 > -- SIP/1001-0000002c is busy > == Everyone is busy/congested at this time (1:1/0/0) > -- Executing [11111111 at incoming-private:4] NoOp("SIP/Proxy-0000002b", > "BUSY") in new stack > -- Executing [11111111 at incoming-private:5] Gosub("SIP/Proxy-0000002b", > "incoming-status,s-BUSY,1") in new stack > > This is what we get on a NO ANSWER call: > --------------------------------------- > -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-0000002f", > "SIP/1001,30") in new stack > == Using SIP RTP CoS mark 5 > == Using SIP VRTP CoS mark 6 > == Using UDPTL CoS mark 5 > -- Called 1001 > -- SIP/1001-00000030 is ringing > -- Nobody picked up in 30000 ms > -- Executing [11111111 at incoming-private:4] NoOp("SIP/Proxy-0000002f", > "NOANSWER") in new stack > -- Executing [11111111 at incoming-private:5] Gosub("SIP/Proxy-0000002f", > "incoming-status,s-NOANSWER,1") in new stack > > This is what we get on a CANCEL call: > ------------------------------------- > -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-00000031", > "SIP/1001,30") in new stack > == Using SIP RTP CoS mark 5 > == Using SIP VRTP CoS mark 6 > == Using UDPTL CoS mark 5 > -- Called 1001 > -- SIP/1001-00000032 is ringing > == Spawn extension (incoming-private, 11111111, 3) exited non-zero on > 'SIP/Proxy-00000031' > > There's no event indicating that a DIALSTATUS is generated and the call > simply doesn't go to the next step in the dialplan. Unless I'm missing > something, it seems to me that it might be a bug. > > I would be happy to get feedback from other users of the DIALSTATUS value > (or Digium), especially in the CANCEL scenario. > > Thank you, > > Michael >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101222/5dd1fc14/attachment.htm>
I see the same thing. Why is there an CANCEL status if it is never set. The only way I have been able to capture a Cancel status is with the h extensions using the 'e' option under dial. But this leaves no way to tell what the DIALSTATUS state was as it is blank. I belive it is a bug as well. Bryant ---------------------------------------- From: "Michael" <voip.question at gmail.com> Sent: Wednesday, December 22, 2010 9:42 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] DIALSTATUS on CANCEL 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> 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' -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101222/1b781492/attachment.htm>
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101222/21c6b47f/attachment.htm>
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:
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asterisk-users mailing list
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Vandar I know understand what you are saying here. Once I turned on CEL I was able to see when and where each hangup was firing for each channel and the order of operations here. I am now moving very aggressively to get to CEL as I now see why CDR's are so broken. I have my CEL to CDR translator in testing and this is looking very promising. Thanks for your help. Bryant ---------------------------------------- From: BryantZ at zktech.com Sent: Friday, December 24, 2010 9:28 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] DIALSTATUS on CANCEL If a call is hung up before an answer our "h" extension is not running in our dial macro Bryant On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan <hvardan71 at gmail.com> wrote:> Hello Bryant > Extension "h" is worked in any case of hangup. > It not important to that the call was answered or no. > It also be more flexible, if you use instead of ${DIALSTATUS}use${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same return code.> http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause > > > -- > 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: >> Vardan >> >> I have not use AEL so it is a bit hard to follow with the formattingthe>> way it is but it looks like correct. >> Please note the "h" extension only appears to run if a call isconnected>> 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 youcan>>> > see, for the other types of reasons, the DIALSTATUS got a value (andwe>>> > 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 everyThurs:>>> > 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 >> >> >> >> -- >> _____________________________________________________________________ >> -- 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/20110101/5290f016/attachment.htm>