predictive not working

[saidmsl]

Hi,

my setup :
vicidialnow 1.3 with astguiclient 2.2rc7
with a trunk sip

dialplan
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${SIP9}/${EXTEN:1},,tTor)
exten => _9X.,3,Hangup

no changes have been made to any files
leads have been uploaded correctly
agents logging in testcampaign

in manual mode : leads appears and agents can talk with leads without problems
in ratio mode : agent do not receive any calls
in asterisk console , no indication of dialing process

any idea? how can i debug ?

Rgds

PS : i'm trying with one agent only

[saidmsl]

Hi,

issue solved.

the problem was :
admin was updated to 2.2 rc7 while the client was still on 2.0.5

[williamconley]

thanks for posting your result (and making it clear to posters that mentioning you are testing the system immediately after an upgrade is Handy information!!)

[saidmsl]

yes
actually, i'm testing vicidial before going to production

it works fine with the latest svn - 2.2. rc7 - with a SIP provider, lime survey integrated

just a comment : the french translation is very bad - need to review it

[mflorell]

Are you volunteering to review the French translation?

We would very much appreciate it since the initial one was done by a non-native speaker, and the updates have been done by babelfish and google translate since then...

[saidmsl]

of course, why not.
if i can have some hints how to start
i will look into the source code

[mflorell]

You don't need to touch the source code, you just need to look at the existing /translations/fr_language.txt and fr_language_admin.txt files included with the 2.2.0rc8 release.

Here is more information on doing translations:
http://wiki.vicidial.org/index.php/VICI:translations

[callingcard]

Would they be the same version? Seems like different versions, which is why it doesn't seem to work ?

What should I check?

I am sorry, just noticed this is VicidialNow forum, and I have a Vicibox installation...


ADMIN : VERSION: 2.2.0-236
BUILD: 100413-2328

Client :
VERSION: 2.2.0-258 BUILD: 100413-1347

Hi,

issue solved.

the problem was :
admin was updated to 2.2 rc7 while the client was still on 2.0.5

[jhonathanm]

hi saidmsl im having the same issue as you did, could you tell me, how did you upgrade the client/agent version to 2.2?

Thank you very much.

[williamconley]

directions for upgrade are included in every download (there is a document named "upgrade" in your /usr/src/astguiclient folder download the NEW version and/or follow the directions for SVN, they can also be read on a Windows Workstation for planning it out, it is very easy)

[charudatta]

My web Client version
VERSION: 2.0.5-206 BUILD: 90525-1014

Asterisk Version : 1.2.30.2

My manual call works fine but when i select Ration pr Adaptive Dialing
in CLI call are dialing but not send on agent ectension

cli output for Manual Dial :
- Executing MeetMe("Local/8600051@default-1e9c,2", "8600051") in new stack
> Channel Local/8600051@default-1e9c,1 was answered.
-- Executing AGI("Local/8600051@default-1e9c,1", "call_log.agi|15088837100") 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/8600051@default-1e9c,1", "SIP/trunk1/15088837100||tTo") in new stack
-- Called trunk1/15088837100
== 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/58600051@default-0d22,2", "call_log.agi|8600051") 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/58600051@default-0d22,2", "SIP/trunk/8600051||tTo") in new stack
-- Called trunk/8600051
-- SIP/trunk-09a6bb88 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("Local/58600051@default-0d22,2", "") in new stack
> Channel Local/58600051@default-0d22,1 was never answered.
== Spawn extension (default, 58600051, 3) exited non-zero on 'Local/58600051@default-0d22,2'
-- Executing DeadAGI("Local/58600051@default-0d22,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
-- SIP/trunk1-099d8820 is making progress passing it to Local/8600051@default-1e9c,1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/trunk1-099d8820 answered Local/8600051@default-1e9c,1

I have hardphone and voipgateway MP124

[williamconley]

when you post, please post your entire configuration including (but not limited to) your installation method, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

Manual call works fine? Do you mean your phone can dial without the server or the phone dials through the server or the agent can log in to a campaign and use the manual dial functions on the agent screen?