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Auto Dial Handoff to Agents Failing

http://forum.eflo.net/VICIDIALforum/viewtopic.php?f=2&t=59

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Auto Dial Handoff to Agents Failing

PostPosted: Wed Jul 05, 2006 2:52 pm
by smt8d1
Hello,

I am new to VICIDIAL, so bear with me. =) So far I have followed the SCRATCH install and am making manual calls (non-autodial) without any problems. As soon as I add in the autodial the call is dropped as soon as it is picked up. This is happening with both the agi-VDADtransfer.agi and the agi-VDADautoREMINDER.agi scripts.

Here is the asterisk log showing the dropped call:
Code: Select all
   -- Registered SIP 'gs102' at 192.168.1.34 port 5060 expires 120
    -- Saved useragent "SJphone/1.60.299a/L (SJ Labs)" for peer gs102
    -- Unregistered SIP 'gs102'
    -- Registered SIP 'gs102' at 192.168.1.34 port 5060 expires 120
    -- Saved useragent "SJphone/1.60.299a/L (SJ Labs)" for peer gs102
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'listencron' logged off from 127.0.0.1
       > Channel SIP/gs102-006a1110 was answered.
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing MeetMe("SIP/gs102-006a1110", "8600051") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
       > Channel SIP/gs102-d016 was answered.
    -- Executing MeetMe("SIP/gs102-d016", "8600051") in new stack
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Saved useragent "SJphone/1.60.299a/L (SJ Labs)" for peer gs102
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/17022203310@default-0dad,2", "call_log.agi|17022203310") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing Dial("Local/17022203310@default-0dad,2", "IAX2/10112:password@72.34.43.5:4569/17022203310||tTo") in new stack
    -- Called 10112:password@72.34.43.5:4569/17022203310
    -- Call accepted by 72.34.43.5 (format gsm)
    -- Format for call is gsm
    -- IAX2/10112-1 is making progress passing it to Local/17022203310@default-0dad,2
    -- IAX2/10112-1 answered Local/17022203310@default-0dad,2
       > Channel Local/17022203310@default-0dad,1 was answered.
    -- Executing AGI("Local/17022203310@default-0dad,1", "call_log.agi|8366") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing DeadAGI("Local/17022203310@default-0dad,2", "call_log.agi|h") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing AGI("IAX2/10112-1", "agi-VDADtransferSURVEY.agi|8366") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing DeadAGI("Local/17022203310@default-0dad,2", "VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----4-----0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
    -- AGI Script agi-VDADtransferSURVEY.agi completed, returning 0
    -- Executing Hangup("IAX2/10112-1", "") in new stack
    -- Executing DeadAGI("IAX2/10112-1", "call_log.agi|h") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script VD_hangup.agi completed, returning 0
    -- AGI Script call_log.agi completed, returning 0
    -- Executing DeadAGI("IAX2/10112-1", "VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
    -- AGI Script VD_hangup.agi completed, returning 0
    -- Hungup 'IAX2/10112-1'


Has anyone else seen this behavior? I have tested the auto-dial handing off to agi-test.agi script and it worked perfectly.

Thanks,
Sean

PostPosted: Wed Jul 05, 2006 4:15 pm
by mflorell
That can be caused by several issues, can you post the Real Asterisk CLI output(asterisk -vvvvvvvvvvvvvvvvvgc NOT asterisk -r)?

Is your IAX trunk registered?
"iax2 show registry"

Reinstall fixed the problem, but I have another issue

PostPosted: Sun Jul 23, 2006 9:04 pm
by smt8d1
Sorry it took so long for me to get back to you. I got pulled away from working on vicidial, and while I was away my vicidial test machine was formatted (don't ask). I went through all of the steps in your SCRATCH INSTALL again and this time the handoff to the SURVEY script worked. I have no idea what I did differently, but I am not complaining. =)

I am now running into a new problem regarding the SURVEY: DTMF tone detection isn't working. Upon picking up the phone the survey message plays, but regardless of what digits I press the script isn't responding. I can't find any entries in the logs indicating that the tones are even being registered.

Do you have any suggestions on where I could look to see if the DTMF tones are being picked up?

Thanks,
Sean

PostPosted: Sun Jul 23, 2006 9:18 pm
by mflorell
What is the call path? (are you calling a local extension)

Can you post some Real Asterisk CLI output?
(asterisk -vvvvvvvvvvvvvvvvgc) not asterisk -r

Debugging Info

PostPosted: Mon Jul 24, 2006 4:09 pm
by smt8d1
mflorell wrote:What is the call path? (are you calling a local extension)

Can you post some Real Asterisk CLI output?
(asterisk -vvvvvvvvvvvvvvvvgc) not asterisk -r



I am calling a POTS line using a VoipJet (IAX2) trunk. The agents are SIP clients using softphones. In the logs below you will see I am just using your gs102/test example.

Code: Select all
   -- Registered SIP 'gs102' at 192.168.1.34 port 5060 expires 120
    -- Saved useragent "SJphone/1.60.299a/L (SJ Labs)" for peer gs102
    -- Unregistered SIP 'gs102'
    -- Registered SIP 'gs102' at 192.168.1.34 port 5060 expires 120
    -- Saved useragent "SJphone/1.60.299a/L (SJ Labs)" for peer gs102
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'listencron' logged off from 127.0.0.1
       > Channel SIP/gs102-006a1110 was answered.
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing MeetMe("SIP/gs102-006a1110", "8600051") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'listencron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'listencron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/17022203310@default-fd4b,2", "call_log.agi|17022203310") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing Dial("Local/17022203310@default-fd4b,2", "IAX2/10112:password@voipjet:4569/17022203310|55|o") in new stack
    -- Called 10112:password@voipjet:4569/17022203310
    -- Call accepted by 64.34.45.100 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/voipjet-3 is making progress passing it to Local/17022203310@default-fd4b,2
    -- IAX2/voipjet-3 answered Local/17022203310@default-fd4b,2
       > Channel Local/17022203310@default-fd4b,1 was answered.
    -- Executing AGI("Local/17022203310@default-fd4b,1", "call_log.agi|8366") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
  == Manager 'sendcron' logged off from 127.0.0.1
  == Spawn extension (default, 17022203310, 2) exited non-zero on 'Local/17022203310@default-fd4b,2'
    -- Executing DeadAGI("Local/17022203310@default-fd4b,2", "call_log.agi|h") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing AGI("IAX2/voipjet-3", "agi-VDADtransferSURVEY.agi|8366") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing DeadAGI("Local/17022203310@default-fd4b,2", "VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----4-----0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
    -- AGI Script VD_hangup.agi completed, returning 0
  == Spawn extension (default, 8366, 2) exited non-zero on 'IAX2/voipjet-3'
    -- Executing DeadAGI("IAX2/voipjet-3", "call_log.agi|h") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
    -- AGI Script call_log.agi completed, returning 0
    -- Executing DeadAGI("IAX2/voipjet-3", "VD_hangup.agi|PRI-----NODEBUG-----0---------------") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
    -- AGI Script VD_hangup.agi completed, returning 0
    -- Hungup 'IAX2/voipjet-3'


As I mentioned before, the survey is administered without any problems, but the survey script isn't picking up any of the digits I am pressing. Could this just be a problem with DTMF detection?

Thanks,
Sean

PostPosted: Mon Jul 24, 2006 4:35 pm
by mflorell
I have never had that kind of problem with IAX trunks, can you try with another carrier? perhaps the free trial of voxee linked on the project site.

It Worked!

PostPosted: Mon Jul 24, 2006 8:06 pm
by smt8d1
You were right. As soon as I switched to Voxee everything worked perfectly. Thanks a lot for debugging that for me!

Sean
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