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Getting Auto-Dialing to Work

PostPosted: Wed Apr 08, 2009 10:06 pm
by rnorm
So I have VicidialNow working with manual calls, but any type of auto-dialing it just wont work. It says I am the only one in the confrence. I setup my 8365 extension. When I login it shows this...

Channel SIP/cc100-097c3bb0 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/cc100-097c3bb0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')


also it keeps showing this...

== 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


Help would be very much appreciated. Thanks!

PostPosted: Thu Apr 09, 2009 3:19 am
by willbordoni
Check your extensions.conf and try to add some Playback(sip-silence) at the beginning of the extensions 8364, 8365, 8366, 8367, 8368, 8369 and 8372.

In facts the correct lines are already there, you can uncomment them.
It gives something like that :

Code: Select all
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDADtransferBROADCAST.agi,${EXTEN})
exten => 8364,4,AGI(agi-VDADtransferBROADCAST.agi,${EXTEN})
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,4,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,6,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup


Hope it will help

Re: Getting Auto-Dialing to Work

PostPosted: Fri Apr 10, 2009 10:06 am
by williamconley
rnorm wrote:but any type of auto-dialing it just wont work.

are you saying that autodial does not dial or autodial does not put the calls into the agent's conference?

are all your screens running? (do some research, use the screen -list command)

PostPosted: Sun Apr 12, 2009 11:38 am
by rnorm
yeah the auto-dial just does not dial i guess the CLI says...

> Channel SIP/cc100-085129b0 was answered.
-- Executing MeetMe("SIP/cc100-085129b0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1


once I pick-up my SIP phone. then it says I'm the only one in the conference. When I setup a Remote agent, it does autodial. I have tried using the SIP-Silence on the 8365 ext and it makes no difference.

by the way... screen -list shows this

There are screens on:
2446.ASTVDadapt (Detached)
2440.ASTVDauto (Detached)
2431.ASTupdate (Detached)
2422.asterisk (Detached)
2443.ASTVDremote (Detached)
2434.ASTsend (Detached)
2449.ASTfastlog (Detached)
2437.ASTlisten (Detached)
8 Sockets in /var/run/screen/S-root.

I am currently using SIP only for my Trunk... not sure if that makes any difference. Any advice would be apprecited. If I can't figure vicidialnow out I will proceed to do the scratch install of vicidial.

Thanks for all your help!

PostPosted: Sun Apr 12, 2009 11:45 am
by rnorm
by the way...

does it matter if I have a Zaptel Card or not? when i load up vicidialnow it says Zaptel : zero chanels configured.

PostPosted: Sun Apr 12, 2009 12:42 pm
by williamconley
ok. progress. the complaint is "does not autodial". autodial requires everything set up properly and then at least one agent available to receive calls (but not already in one) on a campaign with leads to dial. here we have:
rnorm wrote:yeah the auto-dial just does not dial i guess the CLI says...

> Channel SIP/cc100-085129b0 was answered.
-- Executing MeetMe("SIP/cc100-085129b0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1


once I pick-up my SIP phone. then it says I'm the only one in the conference.
This does demonstrate that the agent is logged in, but does not say "available" and does not show the CLI after the client presses "Resume" to become available (in fact, it does not state that the agent pressed resume at all, which would be a possible reason for the system to not dial). So: Did the agent press "resume" and if so, may we see the output from the CLI and the Vicidial Logs during the time the agent pressed "resume"?

AND

When I setup a Remote agent, it does autodial.
If the complaint is that the system does not autodial ... this confuses me somewhat.

PostPosted: Sun Apr 12, 2009 1:01 pm
by rnorm
jesus!!! i feel stupid now. i thought that resume buttom was for recording! sweet thanks!

PostPosted: Sun Apr 12, 2009 9:11 pm
by williamconley
Please read BOTH the Agent manual ... AND the Manager's Manual and read them THOROUGHLY. If you do that, there is no charge for this service. There will be a quiz. (LMAO)

PostPosted: Sun Jun 14, 2009 7:35 pm
by dal
hello..i was confused about extension 8365...it is already in the default extensions.conf right? so, where can i set-up that extension 8365?
i have the same problem with rnorm, manual dialing is working but when i chose ratio it doesn't work even if the agents click already 'resume'...in short autodial doesn't put calls into the agent conference..

hope anyone can help me about this problem...

PostPosted: Sun Jun 14, 2009 10:11 pm
by williamconley
you'll need to open a new thread after you read the stickies to find out what information we need from you.

you don't "set up extension 8365".

read the stickies, then read the manager's manual and go through the exercises one at a time (and don't skip anything) and when you come out the other side, you'll have a functioning system.

the manager's manual is available on eflo.net for FREE for the part you need (although you would do very well to purchase the full version, at least if you consider time to be money).

when you come back, we'll need to know where you had a problem in the manager's manual and your configuration. and start a new thread describing your situation unless it matches this:
Getting Auto-Dialing to Work

So I have VicidialNow working with manual calls, but any type of auto-dialing it just wont work. It says I am the only one in the confrence. I setup my 8365 extension. When I login it shows this...

Channel SIP/cc100-097c3bb0 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/cc100-097c3bb0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')


also it keeps showing this...

== 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


Help would be very much appreciated. Thanks!
and if it DOES match this (as you will note from the guy who opened this thread to begin with) you just need to read the manual because your system is working, you just don't know HOW (which is what the manual is for, right?)