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