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:[email protected]
CfMC http://www.cfmc.com/ On Dec 24, 2010, at 6:03 AM, [email protected] 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 <[email protected]> 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: [email protected] >> 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 >>> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but >>> I may be missing the point on this can anyone else speak to it? >>> >>> Bryant >>> >>> ------------------------------------------------------------------------ >>> *From*: "Vardan Harutyunyan" <[email protected]> >>> *Sent*: Thursday, December 23, 2010 2:11 AM >>> *To*: [email protected] >>> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL >>> >>> I have make test in AEL. >>> >>> context fu { >>> >>> _000./userN => { >>> Dial(SIP/${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: [email protected] >>> www.eif-it.com >>> >>> Bryant Zimmerman wrote: >>>> The Dial Status is not set when accessing it from the h extension. >>>> >>>> Bryant >>>> >>>> ------------------------------------------------------------------------ >>>> *From*: "Vardan Harutyunyan" <[email protected]> >>>> *Sent*: Wednesday, December 22, 2010 10:39 AM >>>> *To*: [email protected] >>>> *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: [email protected] >>>> 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 <[email protected] >>>>> <mailto:[email protected]>> 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
