vicidial.org

by **enjay** » Mon Jun 26, 2006 1:47 pm

I currently have a call into my cell phone and yet I see this in the real-time campaign display.

NO LIVE CALLS WAITING NO AGENTS ON CALLS

Am I a problem child or what..

by **mflorell** » Mon Jun 26, 2006 2:02 pm

are you dialing manual dial or is your dial level in the campaign set to higher than 0?

The realtime campaign campaign display only works in auto dial mode(dial_level > 0).

For manual dial mode you can use the timeoncall report.

by **enjay** » Mon Jun 26, 2006 2:11 pm

That would be it then cause yes Im using manual dial.. I have not yet got into the auto dial mode.. when I set the auto dial level to 4.0 and login and select resume nothing happens..

do I need to restart the cron processes (note theres 2 servers in this config the asterisk server and the apache/db)

-enjay

by **mflorell** » Mon Jun 26, 2006 2:15 pm

The cron processes should be in the crontab exactly as mentioned in the SCRATCH_INSTALL on the asterisk/VICIDIAL server.

The process that controls the dialing in auto-dial mode is AST_VDauto_dial.pl. You should test it by running it manually on your asterisk/VICIDIAL server:

/home/cron/AST_VDauto_dial.pl --debug

by **enjay** » Mon Jun 26, 2006 2:20 pm

while running that this is basically my output

Code: Select all: loop counter: |5759967| TIME DEBUG: -5|-5|0| GMT: 17:18 2006-06-26 12:18:58|LIVE AGENTS LOGGED IN: 0| 2006-06-26 12:18:58| : agents: dial_level: 0| 2006-06-26 12:18:58| : Calls to place: 0 (0 - 0) | 2006-06-26 12:18:58| : CALLING| TIME DEBUG: -5|-5|0| GMT: 17:18 2006-06-26 12:18:58|| lagged call vla agent PAUSED 0|20060626121828|20060626121848|20060626121858|| 2006-06-26 12:18:58|| lagged call vac agent DELETED 0|2006-06-26 12:16:58

if I restart it and relog in it shows 1 agent but no LIVE AGENTS LOGGED IN

Code: Select all: loop counter: |5759999| TIME DEBUG: -5|-5|0| GMT: 17:19 2006-06-26 12:19:54|LIVE AGENTS LOGGED IN: 0| 2006-06-26 12:19:54| : agents: 1 dial_level: 0| 2006-06-26 12:19:54| : Calls to place: 0 (0 - 0) | 2006-06-26 12:19:54| : CALLING| TIME DEBUG: -5|-5|0| GMT: 17:19 2006-06-26 12:19:54|| lagged call vla agent PAUSED 0|20060626121924|20060626121944|20060626121954|| 2006-06-26 12:19:54|| lagged call vac agent DELETED 0|2006-06-26 12:17:54||

after about 30 seconds that agents: 1 goes back to blank.

-enjay

by **mflorell** » Mon Jun 26, 2006 2:33 pm

Is your VICIDIAL server on a separate machine from your database and/or web server?

If so all of them need to be ntp time-synchronized to the same source.

Can you post what the status in the vicidial_live_agents table is for the agent after you click RESUME?

by **enjay** » Mon Jun 26, 2006 2:34 pm

I'll go back and confirm the time source is all correct.

Code: Select all: +---------------+------+-------------+------------+-----------+--------+---------+-------------+-----------+----------+---------+-----------+---------------------+---------------------+---------------------+------------------+----------------+------------+ | live_agent_id | user | server_ip | conf_exten | extension | status | lead_id | campaign_id | uniqueid | callerid | channel | random_id | last_call_time | last_update_time | last_call_finish | closer_campaigns | call_server_ip | user_level | +---------------+------+-------------+------------+-----------+--------+---------+-------------+-----------+----------+---------+-----------+---------------------+---------------------+---------------------+------------------+----------------+------------+ | 13 | 210 | 192.168.0.5 | 8600053 | IAX2/210 | PAUSED | 0 | 32 | 0.0000000 | | | 18638794 | 2006-06-26 12:31:14 | 2006-06-26 12:32:38 | 2006-06-26 12:31:14 | | NULL | 1 | +---------------+------+-------------+------------+-----------+--------+---------+-------------+-----------+----------+---------+-----------+---------------------+---------------------+---------------------+------------------+----------------+------------+

by **enjay** » Mon Jun 26, 2006 2:41 pm

definately a timing issue.. resolved now and appear to be taking calls.. thanks alot matt... more to come

by **lkeller** » Mon Jun 26, 2006 3:59 pm

I am seriously impressed with how much help you've provided in here Matt. You rock dude =)

by **enjay** » Mon Jun 26, 2006 4:02 pm

yea Matt you are a true gentleman and a scholar

by **mflorell** » Mon Jun 26, 2006 4:43 pm

Thank you

I'm glad I can help.

It's usually very easy for me to troubleshoot VICIDIAL since I wrote almost the whole thing myself

by **nekder** » Thu Jan 31, 2008 11:08 am

hello guys
can you please help me,

i have a vey similar problem...

it doesnt dial any call in auodial mode...

i had anther problem before.. in wich vicidial hangups the client at the moment he answers the vicidial auto-dial calls, and the agents doesnt receive any live conference...

then i make some minimal changes, nothing important, and now, it doesnt even
dial the calls...

im so confuse with the autodial, i dont know how it should work...

can you give me some help??
pleeeaseee

thankss

by **enjay** » Thu Jan 31, 2008 1:48 pm

holy old thread..

please provide the typical details

asterisk version
zaptel version
astguiclient version

and anything else you feel appropriate

by **nekder** » Thu Jan 31, 2008 2:40 pm

this are the versions

asterisk 1.2.22
zaptel 1.2.23
astguiclient 2.0.4

im working with a few sip extensións...
mi asterisk is connected with another asterisk throught iax protocol...

this sekond asterisk y linked to a void provider, and to a celulink gsm conversor, for movil calls...

in the first step, i was able to mount calls in the vicidial queue, but only
the movil phone were conected into a live agent session, while the landlines calls were suddenlly hangup by the server at the moment that the client answered them. this were never connected to a live agent session.

id made some minimal changes, and now it doesnt mount calls at all...
i dont exactly remenber this changes, but i can tell that i mess up a litle with the asterisk conf. files (nothing important) and i play with some perl scripts... =S =S

thats all...

thank you =DD

by **nekder** » Thu Jan 31, 2008 4:47 pm

i solved the second problem, i just have to change the campaign, it looks like thera was a missconfiguration problem.

but i still have the firt problem in wich the autodialing doesnt work with the agents... making the calls, but never starting a live session...

the details are the same that i just wrote...

any idea??

thanksss

by **nekder** » Fri Feb 01, 2008 8:39 am

Here i take some CLI outputs

the firts is the call to a landline number (this one doesnt work) and the second is the call to the cellphone number (this one works just fine)...

i note some diferences in the "VDADtransfer.agi" execution...

please take a look...

//*************************LANDLINE********************************************//

-- Executing Dial("Local/0412982912@default-8aa2,2", "iax2/tec:tecnisys@192.168.0.10/0412982912|30|tTo") in new stack
-- Called tec:tecnisys@192.168.0.10/0412982912
-- Call accepted by 192.168.0.10 (format gsm)
-- Format for call is gsm
-- IAX2/desarrollo-5 is making progress passing it to Local/0412982912@default-8aa2,2
-- IAX2/desarrollo-5 answered Local/0412982912@default-8aa2,2
> Channel Local/0412982912@default-8aa2,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/0412982912@default-8aa2,1", "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("Local/0412982912@default-8aa2,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/0412982912@default-8aa2,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/0412982912@default-8aa2,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("Local/0412982912@default-8aa2,1", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on 'Local/0412982912@default-8aa2,1'
-- Executing DeadAGI("Local/0412982912@default-8aa2,1", "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 DeadAGI("Local/0412982912@default-8aa2,1", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
-- Hungup 'IAX2/desarrollo-5'
== Spawn extension (default, 0412982912, 1) exited non-zero on 'Local/0412982912@default-8aa2,2'
-- Executing DeadAGI("Local/0412982912@default-8aa2,2", "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 DeadAGI("Local/0412982912@default-8aa2,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----3)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... --11-----3) completed, returning 0

//***************************CELLPHONE********************************************//

-- Executing Dial("Local/990801392@default-e36b,2", "iax2/tec:tecnisys@192.168.0.10/90801392|30|tTo") in new stack
-- Called tec:tecnisys@192.168.0.10/90801392
-- Call accepted by 192.168.0.10 (format gsm)
-- Format for call is gsm
== 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
-- IAX2/desarrollo-6 answered Local/990801392@default-e36b,2
> Channel Local/990801392@default-e36b,1 was answered.
-- Executing AGI("Local/990801392@default-e36b,1", "agi://127.0.0.1:4577/call_log") in new stack
== Spawn extension (default, 990801392, 1) exited non-zero on 'Local/990801392@default-e36b,2'
-- Executing DeadAGI("Local/990801392@default-e36b,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 AGI("IAX2/desarrollo-6", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/990801392@default-e36b,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----0)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ---3-----0) completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing MeetMe("IAX2/desarrollo-6", "8600051") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/desarrollo-6'
-- Executing DeadAGI("IAX2/desarrollo-6", "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 DeadAGI("IAX2/desarrollo-6", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
-- Hungup 'IAX2/desarrollo-6'

by **mflorell** » Fri Feb 01, 2008 9:30 am

Looks like you need to use the sip-silence line in your dialplan for 8365. The sample extensions.conf shipped with astguiclient has this at the bottom of the file

by **nekder** » Fri Feb 01, 2008 9:55 am

WOOOW!!

you`r a freaking genius

thank you MAtt

it works just fine now =))

i was checking that extension right now,but i never thought that the problem were there...

could you explain how does sip silences work???

thank you again

by **mflorell** » Fri Feb 01, 2008 1:57 pm

It's a problem with SIP and sometimes on IAX where a Local/ channel will not resolve to it's proper channel name pointer for a few seconds which leads to all sorts of problems. The workaround is to play a 1/100 second audio file which for some reason forces channel name resolution immediately. You can thank Chris Doyle for figuring that one out.

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