vicidial.org

Hi Friends,

i am using the vicidialnow. i have a problem with lead list uploading. after uploading the leads it is showing in leads in hopper place on real time screen of campaign. but calls are not landing on agent screen.

i have created user in a right place and even manual dial is working.
this this the version of astguiclient
VERSION: 2.2.1-237
BUILD: 100510-2015


please help me.

thanks
raju

1) Welcome to the Party!

2) You have listed the version of Vicidial with build (VERSION: 2.2.1-237 BUILD: 100510-2015), which is excellent. You have also listed your installer (vicidialNOW). Be aware that there is now a later installer of your distro called GoAutoDial which will install Vicidial 2.2 (which you have) or Vicidial 2.4. And yet another installer called Vicibox (vicibox.com) that will install Vicidial 2.6. At present you are using a fairly old version (2.2.1), and I suggest an upgrade or fresh install with the latest available version of the software for the best experience.

I hope you have read the Vicidial Manager's Manual (available at EFLO.net for free, and there is also a paid version available there with much more in-depth information).

You should also post: 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. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) In your realtime screen, does the system show calls being dialed?

4) In your asterisk command line interface, are there any errors shown?

5) It is possible that you have the wrong IP in your vicidial configuration (if you changed the system IP after building it) or that your carrier is not configured properly or that your carrier is rejecting calls (especially if you do not have any $$ on your account) or that your system is experiencing networking issues, or that your Phone Numbers are not formatted properly, or that your Phone Code is not correct (which is another field on every lead, generally a "1" in the US) ... or any one of several other reasons.

Happy Hunting!

Hi williamconley,


in my real time screen it is showing the calls being dialed:23 and in my asterisk CLI haven't got any errors. as well as i have changed my system ip after installation of vicidial only. now i am making manual dial. but problem is after uploading the leads it is not auto dialing.

we are uploading leads for Australia. please find the lead structure once

phone title first name middle last name address 1 city
xxxxxxxxxx xx xx xx xx xx xx xx

like this. please help me where i was mistaken.

using:
VERSION: 2.2.1-237
BUILD: 100510-2015
Asterisk:1.2.30.2
Single Server
No Digium/Sangoma Hardware
No Extra Software After Installation
Intel Core 2 duo
RAM: 4 GB

If you change the IP address of a Vicidial server, you must run the admin update server ip script. In Vicidial this is located in /usr/share/astguiclient and executed like this:

cd /usr/share/astguiclient
perl /usr/share/astguiclient/ADMIN_update_server_ip.pl

It will then ask you the old and new IP addresses (it should be run one time for each previous IP address that this system may have had, to be safe).

GoAutoDial/VicidialNOW has added a method in their web gui which will allow you to change the IP there, but we don't use that distro of Vicidial (we use Vicibox.com instead, which is a later version).

A reboot is required after this script has completed.

thanks for your suggestion.
but after running this perl /usr/share/astguiclient/ADMIN_update_server_ip.pl script also same problem.

We will likely need to see some asterisk command line output from a single autodial call to see if we can find the problem. (not 3000 lines of code here, just a single call ...)

screen -r
There are several suitable screens on:
2665.asterisk (Detached)
2841.ASTupdate (Detached)
2856.ASTVDadapt (Detached)
2658.astshell20130130101540 (Detached)
2844.ASTsend (Detached)
2847.ASTlisten (Detached)
2850.ASTVDauto (Detached)
2853.ASTVDremote (Detached)
2859.ASTfastlog (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

please help me. i thought every thing is right. but auto dialing was not working

2013-01-30 14:40:59|SERVER CALLS PER SECOND MAXIMUM SET TO: 100 |10||
2013-01-30 14:40:59|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 0|
2013-01-30 14:40:59| : agents: (READY: 0) dial_level: 0|
2013-01-30 14:40:59| : Calls to place: 0 (0 - 0 [0 + 0]) 0 |
2013-01-30 14:40:59|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2013-01-30 14:40:59|LOCAL TRUNK SHORTAGE: 0|0 (0 - 96)|
2013-01-30 14:40:59| : CALLING|

Real-Time Report
Choose Report Display Options
STOP | SLOW | GO MODIFY | SUMMARY
DIAL LEVEL: 4 TRUNK SHORT/FILL: / 0 FILTER: NONE TIME: 2013-01-30 16:03:19
DIALABLE LEADS: 73 CALLS TODAY: 0 AVG AGENTS: 0 DIAL METHOD: RATIO
HOPPER LEVEL: 50 DROPPED / ANSWERED: 0 / 0 DL DIFF: 0 STATUSES: NEW
LEADS IN HOPPER: 50 DROPPED PERCENT: 0% DIFF: 0.00% ORDER: DOWN
+ VIEW MORE VIEW USER GROUP SHOW SERVER INFO HIDE WAITING CALLS SHOW IN-GROUP STATS SHOW PHONES SHOW CUSTPHONES
NO LIVE CALLS WAITING
1 agents logged in 0 agents in calls 1 agents waiting 0 paused agents 0 agents in dead calls

NOTE: i have updated vicidialnow to Vicidial 2.2 SVN. then i got this problem.
this is the real time scree of campaign. here it is showing leads in hopper but not auto dialing. please help me

thanks
raju

But none of this is asterisk command line output showing what happens when a call is made. a single call.

asterisk -R

manual dial call are going.

[Jan 31 09:00:50] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 31 09:00:59] -- Executing AGI("SIP/cc104-08c22890", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 31 09:00:59] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 31 09:00:59] -- Executing Dial("SIP/cc104-08c22890", "SIP/TRUNK/61XXXXXXXXX||tTor") in new stack
[Jan 31 09:00:59] -- Called TRUNK/61XXXXXXXXX
[Jan 31 09:01:01] == Parsing '/etc/asterisk/manager.conf': [Jan 31 09:01:01] Found
[Jan 31 09:01:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 09:01:01] == Parsing '/etc/asterisk/manager.conf': [Jan 31 09:01:01] Found
[Jan 31 09:01:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 09:01:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 09:01:04] -- SIP/TRUNK-08c2c590 is making progress passing it to SIP/cc104-08c22890
[Jan 31 09:01:06] == Parsing '/etc/asterisk/manager.conf': [Jan 31 09:01:06] Found
[Jan 31 09:01:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 09:01:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 09:01:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 09:01:10] -- SIP/TRUNK-08c2c590 answered SIP/cc104-08c22890
[Jan 31 09:01:17] == Spawn extension (default, 961893878822, 2) exited non-zero on 'SIP/cc104-08c22890'
[Jan 31 09:01:17] -- Executing DeadAGI("SIP/cc104-08c22890", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----18-----7") in new stack
[Jan 31 09:01:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --18-----7 completed, returning 0

ok. so how about output from an autodialed call ? is there any?

[Jan 31 02:13:57] -- Executing [61894305050@default:1] AGI("Local/61894305050@default-5cc9,2", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 31 02:13:57] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 31 02:13:57] -- Executing [61894305050@default:2] Dial("Local/61894305050@default-5cc9,2", "SIP/trunk/1894305050|60|tTor") in new stack
[Jan 31 02:13:57] -- Called trunk/1894305050
[Jan 31 02:13:57] == Parsing '/etc/asterisk/manager.conf': [Jan 31 02:13:57] Found
[Jan 31 02:13:57] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 02:13:57] -- Executing [61894305050@default:1] AGI("Local/61894305050@default-fc79,2", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 31 02:13:57] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 31 02:13:57] -- Executing [61894305050@default:2] Dial("Local/61894305050@default-fc79,2", "SIP/trunk/1894305050|60|tTor") in new stack
[Jan 31 02:13:57] -- Called trunk/1894305050
[Jan 31 02:14:01] == Parsing '/etc/asterisk/manager.conf': [Jan 31 02:14:01] Found
[Jan 31 02:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 02:14:01] == Parsing '/etc/asterisk/manager.conf': [Jan 31 02:14:01] Found
[Jan 31 02:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 02:14:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 02:14:03] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 02:14:06] == Parsing '/etc/asterisk/manager.conf': [Jan 31 02:14:06] Found
[Jan 31 02:14:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 02:14:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 31 02:14:34] == Parsing '/etc/asterisk/manager.conf': [Jan 31 02:14:34] Found
[Jan 31 02:14:34] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 31 02:14:34] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8001-00000041'
[Jan 31 02:14:34] -- Executing [h@default:1] DeadAGI("SIP/8001-00000041", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jan 31 02:14:34] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


this is the single call of autodial number. here it is dialed the number but not landing on agent screen.
please help me.

i note that your manual dial calls start with 61

but your autodial calls have the 6 stripped off

what is your dial prefix and manual dial prefix for your campaign and your carrier dial plan?

Also: You did not post your installation method from my earlier request ... very good to list this as it can often link to problems or solutions.