Re: [asterisk-users] Faulty voicemail
I would suggest writing these events into a db in realtime then you can
search through them by the caller id number and piece back together the
call using the unique id. Then you can know exactly what is happening.
Anthony
Adrian Marsh wrote:
> 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:
>>
>>
>>
> "","07x","2000","incomming_pstn","07x","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"
>>
>>
> "","07x","2000","incomming_pstn","07x","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"
>>
>>
> "","07x","2000","incomming_pstn","07x","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"
>>
>>
> "","07x","2000","incomming_pstn","07x","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"
>>
>>
> "","07x","2000","incomming_pstn","07x","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|20070814-085135-07xx-2000-1187077895.3392.WAV")
>> in new stack
>> -- Exe
Re: [asterisk-users] Faulty voicemail
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:
>
>
"","07x","2000","incomming_pstn","07x","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"
>
"","07x","2000","incomming_pstn","07x","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"
>
"","07x","2000","incomming_pstn","07x","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"
>
"","07x","2000","incomming_pstn","07x","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"
>
"","07x","2000","incomming_pstn","07x","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|20070814-085135-07xx-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
>
>
>
> ___
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>
> 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
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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:
>
> "","07x","2000","incomming_pstn","07x","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"
> "","07x","2000","incomming_pstn","07x","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"
> "","07x","2000","incomming_pstn","07x","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"
> "","07x","2000","incomming_pstn","07x","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"
> "","07x","2000","incomming_pstn","07x","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|20070814-085135-07xx-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
>
>
>
> ___
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>
> 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
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[asterisk-users] Faulty voicemail
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:
"","07x","2000","incomming_pstn","07x","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"
"","07x","2000","incomming_pstn","07x","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"
"","07x","2000","incomming_pstn","07x","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"
"","07x","2000","incomming_pstn","07x","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"
"","07x","2000","incomming_pstn","07x","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|20070814-085135-07xx-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
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