vicidial.org

by **smt8d1** » Wed Jul 05, 2006 2:52 pm

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

by **mflorell** » Wed Jul 05, 2006 4:15 pm

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"

by **smt8d1** » Sun Jul 23, 2006 9:04 pm

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

by **mflorell** » Sun Jul 23, 2006 9:18 pm

What is the call path? (are you calling a local extension)

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

by **smt8d1** » Mon Jul 24, 2006 4:09 pm

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

by **mflorell** » Mon Jul 24, 2006 4:35 pm

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.

by **smt8d1** » Mon Jul 24, 2006 8:06 pm

You were right. As soon as I switched to Voxee everything worked perfectly. Thanks a lot for debugging that for me!

Sean

vicidial.org

Auto Dial Handoff to Agents Failing

Auto Dial Handoff to Agents Failing

Reinstall fixed the problem, but I have another issue

Debugging Info

It Worked!

Who is online