[asterisk-users] 1.6.2.0-rc1 intermittent voicemail problem ?

Chris Brookes cbrookes at gmail.com
Tue Sep 15 13:41:54 CDT 2009


1.6.2.0-rc1. I am having trouble with voice mail intermittently not
working correctly on CHANUNAVAIL. (it may happen for other statuses
too, haven't checked). Basically here's what happens:

  -- Executing [1651xxxxxx at mydids:1]
Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack
    -- Executing [s at macro-phone:1] Dial("SIP/ipkall-trunk-14838bc8",
"SIP/1651xxxxxx,25") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Sep 15 14:19:34] WARNING[26239]: app_dial.c:1721 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s at macro-phone:2] Goto("SIP/ipkall-trunk-14838bc8",
"CHANUNAVAIL,1") in new stack
    -- Goto (macro-phone,CHANUNAVAIL,1)
    -- Executing [CHANUNAVAIL at macro-phone:1]
VoiceMail("SIP/ipkall-trunk-14838bc8", "1651xxxxxxx at default,u") in new
stack
    -- <SIP/ipkall-trunk-14838bc8> Playing 'vm-theperson.gsm' (language 'en')
  == Spawn extension (macro-phone, CHANUNAVAIL, 1) exited non-zero on
'SIP/ipkall-trunk-14838bc8' in macro 'phone'
  == Spawn extension (mydids, 165xxxxxx, 1) exited non-zero on
'SIP/ipkall-trunk-14838bc8'

Sometimes it works properly and the extension is read back to the user
and the voicemail is captured. Sometimes it doesn't. So in the failure
scenario, it gets as far as running "Playing 'vm-theperson.gsm'" and
then it just terminates. On the calling party side, the line never
seems to be answered and nothing is heard other than ringing.

The problem seems to occur when the unavailable extension is dialled
multiple times. The first time it works. The second, third, and fourth
time it doesn't'. If you wait 5 minutes it always seems to work on
that first time after an idle duration. It's as though something
hasn't timed out from the first call.

What can I do to try and debug this? Here's the relevant dialplan section:

[sip-inbound-context]
exten => 1651xxxxxxx,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => 1651xxxxxxx,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => 1651xxxxxxx,n,GotoIf($[${CALLERID(num)} = 651xxxxxxx]?:enda)

.. <snipped irrelevant stuff> ..

exten => 1651xxxxxxx,n(enda),Goto(mydids,${DID_EXTEN},1)
exten => 1651xxxxxxx,n,Hangup

[mydids]
exten => 1651xxxxxxx,1,Macro(phone,16513232161)

[macro-phone]
exten => s,1,Dial(SIP/${ARG1},25)
exten => s,n,Goto(${DIALSTATUS},1)
exten => ANSWER,1,Hangup
exten => CANCEL,1,Hangup
exten => NOANSWER,1,Voicemail(${ARG1}@default,u)
exten => BUSY,1,Voicemail(${ARG1}@default,b)
exten => CONGESTION,1,Voicemail(${ARG1}@default,b)
exten => CHANUNAVAIL,1,Voicemail(${ARG1}@default,u)
exten => a,1,VoicemailMain(${ARG1}@default)

Chris



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