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: brya...@zktech.com Sent: Friday, December 24, 2010 9:28 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@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 <hvarda...@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: i...@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 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 >> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but >> I may be missing the point on this can anyone else speak to it? >> >> Bryant >> >> ------------------------------------------------------------------------ >> *From*: "Vardan Harutyunyan" <hvarda...@gmail.com> >> *Sent*: Thursday, December 23, 2010 2:11 AM >> *To*: asterisk-users@lists.digium.com >> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL >> >> I have make test in AEL. >> >> context fu { >> >> _000./userN => { >> Dial(SIP/${EXTEN:3...@prov); >> Noop(${DIALSTATUS}); >> }; >> h => { >> Noop(${DIALSTATUS}); >> }; >> }; >> >> And look CLI >> -- Executing [00018185402...@fu:1] NoOp("SIP/userN-b6317738", "") >> in new stack >> -- Executing [00018185402...@fo:2] Dial("SIP/user3-b6317738", >> "SIP/18185402...@prov") in new stack >> -- Called 18185402...@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 [...@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: i...@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" <hvarda...@gmail.com> >>> *Sent*: Wednesday, December 22, 2010 10:39 AM >>> *To*: asterisk-users@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: i...@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.nik...@cem-solutions..net >>> > <mailto:d.nik...@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 >> >> >> >> -- >> _____________________________________________________________________ >> -- 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