<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Vandar<br />
<br />
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.<br />
<br />
Thanks for your help.<br />
Bryant<br />
<br />
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<span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"><hr width="100%" size="2" align="center" />
<b>From</b>: BryantZ@zktech.com<br />
<b>Sent</b>: Friday, December 24, 2010 9:28 AM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] DIALSTATUS on CANCEL</span><br />
<br />
If a call is hung up before an answer our "h" extension is not running in our dial macro <br />
<br />
Bryant<br />
<br />
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan <hvardan71@gmail.com> wrote:<br />
<br />
> Hello Bryant<br />
> Extension "h" is worked in any case of hangup.<br />
> It not important to that the call was answered or no.<br />
> 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.<br />
> http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause<br />
> <br />
> <br />
> -- <br />
> Vardan Harutyunyan,<br />
> Senior System Administrator<br />
> <br />
> Enterprise Incubator Foundation<br />
> 123 Hovsep Emin Street,<br />
> Yerevan 0051, Republic of Armenia<br />
> Tel: + 374 10 219735<br />
> Fax: + 374 10 219777<br />
> E-mail: info@eif.am<br />
> www.eif-it.com<br />
> <br />
> Bryant Zimmerman wrote:<br />
>> Vardan<br />
>> <br />
>> I have not use AEL so it is a bit hard to follow with the formatting the<br />
>> way it is but it looks like correct.<br />
>> Please note the "h" extension only appears to run if a call is connected<br />
>> so I do not know when the "CANCEL" would ever be set.<br />
>> There may be someone else who can speak to this. It also appears thet<br />
>> ${DIALSTATUS} may not be set if the call is not allowed to time out or<br />
>> dialed. To me it would make sense to set the inital state of the<br />
>> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but<br />
>> I may be missing the point on this can anyone else speak to it?<br />
>> <br />
>> Bryant<br />
>> <br />
>> ------------------------------------------------------------------------<br />
>> *From*: "Vardan Harutyunyan" <hvardan71@gmail.com><br />
>> *Sent*: Thursday, December 23, 2010 2:11 AM<br />
>> *To*: asterisk-users@lists.digium.com<br />
>> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL<br />
>> <br />
>> I have make test in AEL.<br />
>> <br />
>> context fu {<br />
>> <br />
>> _000./userN => {<br />
>> Dial(SIP/${EXTEN:3}@Prov);<br />
>> Noop(${DIALSTATUS});<br />
>> };<br />
>> h => {<br />
>> Noop(${DIALSTATUS});<br />
>> };<br />
>> };<br />
>> <br />
>> And look CLI<br />
>> -- Executing [00018185402020@fu:1] NoOp("SIP/userN-b6317738", "")<br />
>> in new stack<br />
>> -- Executing [00018185402020@fo:2] Dial("SIP/user3-b6317738",<br />
>> "SIP/18185402020@Prov") in new stack<br />
>> -- Called 18185402020@Prov<br />
>> -- SIP/Prov-082a83b8 is making progress passing it to<br />
>> SIP/userN-b6317738<br />
>> == Spawn extension (fu, 00018185402020, 2) exited non-zero on<br />
>> 'SIP/user3-b6317738'<br />
>> -- Executing [h@fu:1] NoOp("SIP/userN-b6317738", "CANCEL") in new stack<br />
>> <br />
>> I think, I am right<br />
>> <br />
>> --<br />
>> Vardan Harutyunyan,<br />
>> Senior System Administrator<br />
>> <br />
>> Enterprise Incubator Foundation<br />
>> 123 Hovsep Emin Street,<br />
>> Yerevan 0051, Republic of Armenia<br />
>> Tel: + 374 10 219735<br />
>> Fax: + 374 10 219777<br />
>> E-mail: info@eif.am<br />
>> www.eif-it.com<br />
>> <br />
>> Bryant Zimmerman wrote:<br />
>>> The Dial Status is not set when accessing it from the h extension.<br />
>>> <br />
>>> Bryant<br />
>>> <br />
>>> ------------------------------------------------------------------------<br />
>>> *From*: "Vardan Harutyunyan" <hvardan71@gmail.com><br />
>>> *Sent*: Wednesday, December 22, 2010 10:39 AM<br />
>>> *To*: asterisk-users@lists.digium.com<br />
>>> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL<br />
>>> <br />
>>> Try to use h extension<br />
>>> <br />
>>> --<br />
>>> Vardan Harutyunyan,<br />
>>> Senior System Administrator<br />
>>> <br />
>>> Enterprise Incubator Foundation<br />
>>> 123 Hovsep Emin Street,<br />
>>> Yerevan 0051, Republic of Armenia<br />
>>> Tel: + 374 10 219735<br />
>>> Fax: + 374 10 219777<br />
>>> E-mail: info@eif.am<br />
>>> www.eif-it.com<br />
>>> <br />
>>> Michael wrote:<br />
>>> > Hi Nikhil,<br />
>>> ><br />
>>> > Both debug and verbose are set to 20. That's all I got, but as you can<br />
>>> > see, for the other types of reasons, the DIALSTATUS got a value (and we<br />
>>> > see the events). I'm pretty sure it's a bug.<br />
>>> ><br />
>>> > Michael<br />
>>> ><br />
>>> > On Wed, Dec 22, 2010 at 9:01 AM, Nikhil <d.nikhil@cem-solutions..net<br />
>>> > <mailto:d.nikhil@cem-solutions.net>> wrote:<br />
>>> ><br />
>>> > Hi<br />
>>> > Enable debug level to more than 1 ,you may get something.<br />
>>> ><br />
>>> > Thanks<br />
>>> > Nikhil<br />
>>> ><br />
>>> > On 12/22/2010 11:26 AM, Michael wrote:<br />
>>> ><br />
>>> > Spawn extension (incoming-private, 11111111, 3) exited non-zero<br />
>>> > on 'SIP/Proxy-00000031'<br />
>>> ><br />
>>> ><br />
>>> ><br />
>>> ><br />
>>> > --<br />
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>> <br />
>> <br />
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