Enable to find agi-VDADtransferSURVEY.agi

[jack_decent]

i have install from iso:
GoAutoDial CE 2.0
Vicidial 2.2.1
Asterisk 1.4.27.1-vici
VERSION: 2.2.1-260 BUILD: 100527-2211

i want to run servey campaign but i m enable to find /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY.agi
There is no file named agi-VDADtransferSURVEY.agi

i read manager manual but it hard to understand so plz guide how to creat servey campaign easily way .

Any help appreciated

[Michael_N]

look for agi-VDAD_ALL_outbound

[williamconley]

Thank you for listing your installer and vicidial version with build!!!

Now: Read the Manual!! *Free version on EFLO.net* (I do highly recommend the paid version, especially for functions like survey!)

Here's where you'll find the settings, then you can read the manual and the (?) links to get the jist of it:

Campaigns->[choose yours]->Detail view (top menu)->Survey

DO NOT CHANGE ANYTHING!! First: get it working with the default settings (this is done by changing the campaign VDAD Exten to 8366). AFTER you have it working, change one thing at a time and verify you didn't just screw it up before moving on.

I mean it! Don't change anything, not even the recordings, until you have made it work once!!!

[jack_decent]

Thanks willaim its really help full
I change my VDAD ext 8366 but no survey run its work like 8369 no survey is running .

[williamconley]

I change my VDAD ext 8366

Under Campaigns->[choose yours]

you changed "Campaign VDAD exten:" to 8366 ... and then you logged in as an agent and the prospects did not hear the survey?

Please post your CLI output from this example (not from any other events in the system, it should be idle except for this test)

[jack_decent]

you changed "Campaign VDAD exten:" to 8366 ... and then you logged in as an agent and the prospects did not hear the survey?

yes willaim exactly ..

Plz see my logs and reply soon
Thanks in Advance

Created MeetMe conference 1023 for conference '8600052'
[Feb 7 11:20:31] -- <SIP/cc100-0000003d> Playing 'conf-onlyperson' (language 'en')
[Feb 7 11:20:34] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 11:20:43] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:20:43] Found
[Feb 7 11:20:43] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:20:43] -- Executing [8600052@default:1] MeetMe("Local/8600052@default-be5b,2", "8600052|F") in new stack
[Feb 7 11:20:43] > Channel Local/8600052@default-be5b,1 was answered.
[Feb 7 11:20:43] -- Executing [919727824200@default:1] AGI("Local/8600052@default-be5b,1", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 7 11:20:43] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 7 11:20:43] -- Executing [919727824200@default:2] Dial("Local/8600052@default-be5b,1", "SIP/19727824200@XXXXX.com||tTor") in new stack
[Feb 7 11:20:43] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:20:43] Found
[Feb 7 11:20:43] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:20:43] -- Executing [58600052@default:1] MeetMe("Local/58600052@default-3e5a,2", "8600052|Fmq") in new stack
[Feb 7 11:20:43] > Channel Local/58600052@default-3e5a,1 was answered.
[Feb 7 11:20:43] -- Executing [8309@default:1] Answer("Local/58600052@default-3e5a,1", "") in new stack
[Feb 7 11:20:43] -- Executing [8309@default:2] Monitor("Local/58600052@default-3e5a,1", "wav|20110207-212042_9727824200") in new stack
[Feb 7 11:20:43] -- Executing [8309@default:3] Wait("Local/58600052@default-3e5a,1", "3600") in new stack
[Feb 7 11:20:43] -- ast_get_srv: SRV lookup for '_sip._udp.XXXXX.com' mapped to host sip1.XXXXX.com, port 5060
[Feb 7 11:20:43] -- Called 19727824200@XXXXX.com
[Feb 7 11:20:45] -- SIP/XXXXX.com-0000003e is making progress passing it to Local/8600052@default-be5b,1
[Feb 7 11:20:47] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 11:20:47] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 11:20:48] -- SIP/XXXXX.com-0000003e answered Local/8600052@default-be5b,1
[Feb 7 11:20:52] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:20:52] Found
[Feb 7 11:20:52] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:20:52] -- Executing [h@default:1] DeadAGI("Local/8600052@default-be5b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----4") in new stack
[Feb 7 11:20:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---9-----4 completed, returning 0
[Feb 7 11:20:52] == Spawn extension (default, 919727824200, 2) exited non-zero on 'Local/8600052@default-be5b,1'
[Feb 7 11:20:52] == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-be5b,2'
[Feb 7 11:20:52] -- Executing [h@default:1] DeadAGI("Local/8600052@default-be5b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Feb 7 11:20:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 7 11:20:52] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:20:52] Found
[Feb 7 11:20:52] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:20:52] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:20:52] Found
[Feb 7 11:20:52] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:20:52] == Spawn extension (default, 58600052, 1) exited non-zero on 'Local/58600052@default-3e5a,2'
[Feb 7 11:20:52] -- Executing [h@default:1] DeadAGI("Local/58600052@default-3e5a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Feb 7 11:20:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 7 11:20:52] == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600052@default-3e5a,1'
[Feb 7 11:20:52] -- Executing [h@default:1] DeadAGI("Local/58600052@default-3e5a,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Feb 7 11:20:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
7 11:21:00] == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/cc100-0000003d'
[Feb 7 11:21:00] -- Executing [h@default:1] DeadAGI("SIP/cc100-0000003d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Feb 7 11:21:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 7 11:21:00] == Parsing '/etc/asterisk/manager.conf': [Feb 7 11:21:00] Found
[Feb 7 11:21:00] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:21:00] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-acf1,2", "8600052|K") in new stack
[Feb 7 11:21:00] WARNING[6770]: app_meetme.c:2985 admin_exec: Conference number '8600052' not found!
[Feb 7 11:21:00] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-acf1,2", "") in new stack
[Feb 7 11:21:00] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-acf1,2'
[Feb 7 11:21:00] -- Executing [h@default:1] DeadAGI("Local/55558600052@default-acf1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Feb 7 11:21:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[

[williamconley]

is this a manual dial campaign or an autodial campaign?

[jack_decent]

For cheking purpose i m set it to manual dial.
but when survey settings done we want to run it on ratio or average.

[williamconley]

you cannot do survey campaigns in manual. try your test in Ratio mode only.

[jack_decent]

Thanks william its working on ratio mode but can't understand how to change this script.
1-DEMOCRAT|2-REPUBLICAN|3-INDEPENDANT|8-OPTOUT|X-NO RESPONSE|

i want first run hello.gsm then press 1 for intrested or press 2 for not intrested then thanks.gsm.Plz guide me how to configure

[williamconley]

Thanks william its working on ratio mode but can't understand how to change this script.
1-DEMOCRAT|2-REPUBLICAN|3-INDEPENDANT|8-OPTOUT|X-NO RESPONSE|

WHY do you want to change those values?

have you gone through the manual for setting up a survey? (step by step, skipping nothing, altering nothing ... just following the manual?)

[jack_decent]

Thanks william now i m ready to start my survey campaign .I m really thank full to u.
Few more question
In survey campaign auto answering machine detection is active or not ??
How and where we record conversation on this campaign?
if user manual dial directly from xlite how to record this call?
We want to play music when agent wait for we want to play some song when he waiting for next call.is it possible?

[williamconley]

Thanks william now i m ready to start my survey campaign .I m really thank full to u.
Few more question
In survey campaign auto answering machine detection is active or not ??
How and where we record conversation on this campaign?

Before we tell you everything that's in the manual ... have you READ the manual? (eflo.net -> free download, but i strongly recommend the PAID version, as it has hundreds of pages of TIME SAVING directions, walk-throughs, and hints).

if user manual dial directly from xlite how to record this call?
We want to play music when agent wait for we want to play some song when he waiting for next call.is it possible?

I do not advise allowing users to dial directly from xlite. Require them to log in to a manual dial campaign. Then you can not only record the call, but also capture the lead information or just record the lead information on an existing record. If you were to record the call WITHOUT logging in to Vicidial ... how would you manage those recordings? You don't have the call linked to a record in Vicidial ... who will administer this new call recording system? You'd have to build ALL that. Pointless.

MOH can be added by a fairly simple alteration to the meetme room initialization, look at the extensions.conf file and look up the "meetme" command in asterisk. You'll need to find the appropriate line and add one letter (I think it's "M", if i recall ).

Code: Select all

exten => _86000[5-9]X,1,Meetme,${EXTEN}|FM
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|FM

[jack_decent]

hello william i have a problem with my survey campaign
call not transfer to my agent
calls waiting for agents shows 6 calls waiting for agent but call not transfer to my agent..
Plz rep fast

[williamconley]

Show CLI from a SINGLE call on a TEST campaign from the moment of "dial" to the moment of hangup. Do not have any other traffic on the system.

And once again: Have you READ the manual? Have you run a TEST survey campaign before attempting to launch a real one? (Successfully, of course!)

[jack_decent]

i do check before its working fine on testing or my pc but problem with agents.
i m really confuse ..

[jack_decent]

This is last Call Logs

-- Executing AGI("Local/914164820836@default-d71e,2", "agi://127.0.0.1:4577/call_log") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919054630161@default-871d,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/914164820836@default-d71e,2", "SIP/14164820836@xxxvoip.com||tTor") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/919054630161@default-871d,2", "SIP/19054630161@xxxvoip.com||tTor") in new stack
-- parse_srv: SRV mapped to host sip2.xxxvoip.com, port 5060
-- parse_srv: SRV mapped to host sip2.xxxvoip.com, port 5060
-- Called 14164820836@xxxvoip.com
-- Called 19054630161@xxxvoip.com
-- SIP/xxxvoip.com-08778500 is making progress passing it to Local/914164820836@default-d71e,2
Feb 9 11:32:28 NOTICE[5254]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 70.42.75.4
-- SIP/xxxvoip.com-0878abd8 is ringing
-- SIP/xxxvoip.com-0878abd8 is making progress passing it to Local/919054630161@default-871d,2
-- SIP/xxxvoip.com-0878abd8 answered Local/919054630161@default-871d,2
> Channel Local/919054630161@default-871d,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/919054630161@default-871d,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 919054630161, 2) exited non-zero on 'Local/919054630161@default-871d,2'
-- Executing DeadAGI("Local/919054630161@default-871d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----0 completed, returning 0
-- Executing AGI("SIP/xxxvoip.com-0878abd8", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("SIP/xxxvoip.com-0878abd8", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi

[williamconley]

you will need to check your agi logs and/or log back in on YOUR pc and find what the difference is between your login and theirs.

For instance: did they ... resume?

[jack_decent]

How do i check agi logs and/or log back in on my pc ??

[jack_decent]

William i fell the actual problem is there.

{ [Feb 9 12:26:32] NOTICE[8987]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '6472881289' rejected because extension not found. }

or its fail to launch agi-VDAD_ALL_outbound.agi

-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi

because its stuck on here and then report shows 1 call waiting for agent ..


AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8366@default:3] AGI("SIP/xxx.com-0000002b", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'LiveMsgdebt1' (escape_digits=123456789) (sample_offset 0)
[Feb 9 12:26:32] NOTICE[8987]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '6472881289' rejected because extension not found.
== Refreshing DNS lookups.
[Feb 9 12:26:51] NOTICE[8987]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '6472881289' rejected because extension not found.

plz suggest what can i do ??

[williamconley]

since your login worked, but the agents don't ... go back and test and find the difference (first make sure yours still works.)

[jack_decent]

yes its surelly working on 8369 this problem appears only 8366..

[williamconley]

if you are saying that it works on 8369 but not on 8366, you have not completed the setup of the survey in a test yet. Go back to the manual and set up a survey campaign step by step. Read that first post I made, start over, do it right. If you skip steps you could be at this forever

If you do it one step at a time, you could get it right today.

That's why Matt made the manual. That's why the default Survey setup is there.

If you have a problem with the default setup while working the tutorial in the manual, ask for help and we'll walk you through it. Since we know all the components. Until you've done that, we have no way of knowing what else could be wrong, so do that first.

[jack_decent]

Its working
Thx Dude

[williamconley]

Now ... to thank me, of course, you come back here and walk the next 5 newbies through this same "just follow the manual, use the example, go slow ..." procedure I took you through

[jack_decent]

MOH can be added by a fairly simple alteration to the meetme room initialization, look at the extensions.conf file and look up the "meetme" command in asterisk. You'll need to find the appropriate line and add one letter (I think it's "M", if i recall ..

exten => _86000[5-9]X,1,Meetme,${EXTEN}|FM
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|FM

I m add this M in extensions.conf but now where upload my music files.

[williamconley]

MOH is a function of Asterisk itself. There are also methods built into Vicidial (have you activated the audio store?). Have you read the manual?