Hmm... He swears he heard a voice saying he'd dialed the number incorrectly.. But that's no-where in the dialplan, and I do see the incoming calls correctly for the times he's saying..
Adrian Marsh -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: 14 August 2007 15:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faulty voicemail Adrian Marsh wrote: > Hi All, > > I was made aware today that some of my calls coming in are not going to > voicemail... Below are some logs, and the macro that should run on the > incoming_pstn context for that extension. I can see that theres a > non-zero exit before it gets to voicemail, but I've no idea why. In > this case theres 2 SIP clients to sim-call. On other occasions it works > fine. In the CDR logs, I can see "NO ANSWER" and "ANSWERED" - what > would be there if voicemail "answers"? > > Asterisk: 1.2.23 > > [macro-ext-group-home] > ; ${ARG1} - Virtual Extension (e.g. 2005) > exten => > s,1,ExecIF($["${RECORDSIP}"="TRUE"],Monitor,wav|${TIMESTAMP}-${CALLERID( > num)}-${MACRO_EXTEN}-${UNIQUEID}.WAV) > exten => > s,2,Dial(SIP/2${ARG1:-2}&SIP/4${ARG1:-2}&SIP/6${ARG1:-2},${OFFICE_TIMEOU > T},rw) > exten => s,3,Voicemail(u${ARG1}) > exten => s,103,Voicemail(u${ARG1}) > > > The call logs show: > > "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-1 > ","SIP/600-08e0b990","Dial","SIP/200&SIP/400&SIP/600|15|rw","2007-08-14 > 08:49:16",,"2007-08-14 08:49:18",2,0,"NO ANSWER","DOCUMENTATION" > "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-2 > ","SIP/600-08e19d58","Dial","SIP/200&SIP/400&SIP/600|15|rw","2007-08-14 > 08:49:46",,"2007-08-14 08:49:56",10,0,"NO ANSWER","DOCUMENTATION" > "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-1 > ","SIP/600-08e0b990","VoiceMail","u2000","2007-08-14 > 08:50:37","2007-08-14 08:50:52","2007-08-14 > 08:51:00",23,8,"ANSWERED","DOCUMENTATION" > "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-2 > ","SIP/600-08e19d58","Dial","SIP/200&SIP/400&SIP/600|15|rw","2007-08-14 > 08:51:35",,"2007-08-14 08:51:45",10,0,"NO ANSWER","DOCUMENTATION" > "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-1 > ","SIP/600-08e0b990","VoiceMail","u2000","2007-08-14 > 08:52:19","2007-08-14 08:52:34","2007-08-14 > 08:52:38",19,4,"ANSWERED","DOCUMENTATION" > > And my messages log for that time (for one failed call) shows: > > ubiphone*CLI> > -- Accepting AUTHENTICATED call from 193.111.200.135: > > requested format = alaw, > > requested prefs = (), > > actual format = ulaw, > > host prefs = (ulaw|alaw), > > priority = mine > > ubiphone*CLI> > -- Executing Macro("IAX2/ubigradin-2", "ext-group-home|2000") in new > stack > -- Executing ExecIf("IAX2/ubigradin-2", > "0|Monitor|wav|20070814-085135-07xxxxxxxxxx-2000-1187077895.3392.WAV") > in new stack > -- Executing Dial("IAX2/ubigradin-2", > "SIP/200&SIP/400&SIP/600|15|rw") in new stack > > ubiphone*CLI> > -- Called 200 > Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable to > create channel of type 'SIP' (cause 3 - No route to destination) > -- Called 600 > > ubiphone*CLI> > -- SIP/600-08e19d58 is ringing > > ubiphone*CLI> > -- SIP/200-08e0b990 is ringing > > ubiphone*CLI> > == Spawn extension (macro-ext-group-home, s, 2) exited non-zero on > 'IAX2/ubigradin-2' in macro 'ext-group-home' > == Spawn extension (macro-ext-group-home, s, 2) exited non-zero on > 'IAX2/ubigradin-2' > -- Hungup 'IAX2/ubigradin-2' > > > > Adrian Marsh > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Exited non-zero here looks like the caller hungup before going to voicemail, the caller is the one thing you can't control. Anthony _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
