I am using the g option and it does not run the next statement or "h" extension if the caller hangs up before an answers or time out event occurs during a dial comand.
Bryant On Dec 24, 2010, at 9:55 AM, Jim Dickenson <dicken...@cfmc.com> wrote: > If on the dial command you add option g, if the call is not answered, it will > fall through to the next statement which can be a hangup command and then it > will go to the h extension. If that does not then make the statement after > the dial command a goto h extension. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Dec 24, 2010, at 6:03 AM, brya...@zktech.com wrote: > >> 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 -- _____________________________________________________________________ -- 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
