Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 16:48:28 -0700 (PDT), Steve Edwards asterisk@sedwards.com wrote: Somehow, I'm guessing that 'failed' means that something failed while processing the call file or that the call failed to answer, not that somebody terminated the call. Thanks guys. After testing with a PCI card + Dahdi, and then with a Linksys 3102, turns out that neither jumps to the failed or h extension when the remote number is busy, ie. already engaged (with no support for callwaiting, ie. two-way calling) == extensions.conf [internal] ;call from XLite ;exten = _5.,1,Dial(Dahdi/1/${EXTEN}) exten = _5.,1,Dial(SIP/3102-fxo/${EXTEN}) exten = h,1,NoOp(Called ended with ${DIALSTATUS}) exten = failed,1,NoOp(Call ended with ${REASON}) == CLI == Using SIP RTP CoS mark 5 -- Executing [5551234@internal:1] Dial(SIP/xlite-000e, SIP/3102-fxo/5551234) in new stack == Using SIP RTP CoS mark 5 -- Called 3102-fxo/5551234 #Here, phone is still ringing, but Asterisk wrongly says it has answered -- SIP/3102-fxo-000f is ringing -- SIP/3102-fxo-000f answered SIP/xlite-000e #Says it bridged calls although remote end hasn't answered -- Packet2Packet bridging SIP/xlite-000e and SIP/3102-fxo-000f == As I no longer have a real landline, it could be due to the way my ADSL VoIP landline works. Bottom line: I can't use that line to write a robocall. Thanks guys. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Friday, March 18, 2011 10:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas da...@debsinc.com wrote: exten = start,n,Playback(manolo_camp-morning_coffee) ;exten = start,n,Hangup() exten = start,n,Goto(${EXTEN}-${REASON}) ;not run ;exten = failed,1,NoOp(Call ended with ${REASON}) ;not run ;exten = s,1,NoOp(Call ended with ${REASON}) ;empty ;exten = h,1,NoOp(Call ended with ${REASON}) ;not run exten = start-NOANSWER,1,NoOp(Call ended with ${REASON}) === Is this what you had in mind? Thank you. That's the ticket. Unfortunately, it can only jump to h, and ${REASON} is empty. Based on... www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example ... I also tried this, but Asterisk doesn't jump to any of those extensions: = extensions.conf ... exten = start,n,Playback(manolo_camp-morning_coffee) ;exten = start,n,Hangup() ;exten = start,n,Goto(${EXTEN}-${REASON}) exten = start,n,Goto(s-${DIALSTATUS},1) exten = s-ANSWER,1,Hangup exten = s-CANCEL,1,Hangup exten = s-NOANSWER,1,Hangup exten = s-BUSY,1,Busy ;Only works with SIP calls exten = s-CHANUNAVAIL,1,Verbose(Not available) exten = s-CONGESTION,1,Congestion exten = _s-.,1,Congestion exten = s-,1,Congestion = CLI -- Executing [start@callback:5] Playback(DAHDI/1-1, manolo_camp-morning_coffee) in new stack -- DAHDI/1-1 Playing 'manolo_camp-morning_coffee.ulaw' (language 'fr') == Spawn extension (callback, start, 5) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' [Mar 18 16:41:35] NOTICE[1200]: pbx_spool.c:349 attempt_thread: Call completed to Dahdi/1/5551234 = Is there no way to know how a call ended? Thank you. I believe you will achieve the desired result by replacing ${REASON} with ${HANGUP_CAUSE}. -- _ -- 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Behalf Of Gilles Unfortunately, it can only jump to h, and ${REASON} is empty. On Fri, 18 Mar 2011, Danny Nicholas wrote: I believe you will achieve the desired result by replacing ${REASON} with ${HANGUP_CAUSE}. REASON is documented as being valid in the 'failed' extension. If it is not working as you expect it to, maybe you could read through the source (/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why. You could always submit a patch... HANGUP_CAUSE should be HANGUPCAUSE. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards asterisk@sedwards.com wrote: On Fri, 18 Mar 2011, Danny Nicholas wrote: I believe you will achieve the desired result by replacing ${REASON} with ${HANGUP_CAUSE}. REASON is documented as being valid in the 'failed' extension. If it is not working as you expect it to, maybe you could read through the source (/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why. You could always submit a patch... HANGUP_CAUSE should be HANGUPCAUSE. Thanks guys. In which case does Asterisk jump to the failed extension? -- _ -- 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On 03/18/2011 05:43 PM, Gilles wrote: On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards asterisk@sedwards.com wrote: On Fri, 18 Mar 2011, Danny Nicholas wrote: I believe you will achieve the desired result by replacing ${REASON} with ${HANGUP_CAUSE}. REASON is documented as being valid in the 'failed' extension. If it is not working as you expect it to, maybe you could read through the source (/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why. You could always submit a patch... HANGUP_CAUSE should be HANGUPCAUSE. Thanks guys. In which case does Asterisk jump to the failed extension? You need to define the 'failed' extension in your context to have the ${REASON} variable set (I've found). exten = failed,1,NoOp(Failure reason is: ${REASON}) -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 17:56:12 -0500, Anthony Messina amess...@messinet.com wrote: You need to define the 'failed' extension in your context to have the ${REASON} variable set (I've found). exten = failed,1,NoOp(Failure reason is: ${REASON}) Thanks but for some reason, after calling out through a call file, Asterisk doesn't jump to it although the callee hangs up while Asterisk is still playing: === [callback] exten = start,1,Wait(2) exten = start,n,ChanIsAvail(Dahdi/1) exten = start,n,NoOp(${AVAILORIGCHAN})}) exten = start,n,Answer() exten = start,n,Playback(manolo_camp-morning_coffee) ;exten = start,n,Hangup() ;not run exten = failed,1,NoOp(Call ended with ${REASON}) === -- _ -- 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Sat, 19 Mar 2011, Gilles wrote: Thanks but for some reason, after calling out through a call file, Asterisk doesn't jump to it although the callee hangs up while Asterisk is still playing: Somehow, I'm guessing that 'failed' means that something failed while processing the call file or that the call failed to answer, not that somebody terminated the call. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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