Agent cannot login

[john_usc]

I just downloaded vicidialnow1.1 and I followed the managers manual to get the campaigns and everything set up but when I go to the
http://serveradd/agc/vicidial.php

and I enter the phone login and password and then it asks me for user info so I enter user 7777 with password test and with the campaign from drop down but on submit it gives me this error

Sorry, your phone login and password are not active in this system, please try again:


can anyone please help!!!!
Thanks

[john_usc]

it worked. the login was 100 not cc100. my bad
However the bigger problem id that when I click the maual dial I get lots of popups and the system says its waiting for dialing and some seconds in popup box...I have to restart the browser or it keeps poping up

What could be wrong

[john_usc]

this is what it says the first time I click
DIAL NEXT NUMBER icon

"Calling: (720)333-1234 UID: M0102001001000000004   Waiting for Ring.."

The above pops up with an option of ok. When I click ok then this pop ups

"
Calling: (720)333-1234 UID: M0102001001000000004   Waiting for Ring... 1 seconds
"

and the more I hit ok it just keep coming and after 4th ok it shows this


"Calling: <span isdynflag="1" info="Call +17203331234;0;+17203331234;1;" onmouseup="SkypeSetCallButtonPressed(this, 0,0,0)" onmousedown="SkypeSetCallButtonPressed(this, 1,0,0)" onmouseover="SkypeSetCallButton(this, 1,0,0);skype_active=SkypeCheckCallButton(this);" onmouseout="SkypeSetCallButton(this, 0,0,0);HideSkypeMenu();" context="(720)333-4828" reallyisdynflag="1" fax="0" rtl="false" class="skype_tb_injection" id="__skype_highlight_id"><span title="Skype actions" onmouseout="SkypeSetCallButtonPart(this, 0);" onmouseover="SkypeSetCallButtonPart(this, 1);" class="skype_tb_injection_left" id="__skype_highlight_id_left"><span style="background-image: url(chrome://skype_ff_toolbar_win/content/cb_normal_l.gif);" class="skype_tb_injection_left_img" id="__skype_highlight_id_left_adge"><img src="chrome://skype_ff_toolbar_win/content/cb_transparent_l.gif" style="height: 11px; width: 7px;" class="skype_tb_img_adge" height="11"></span><span class="skype_tb_injection_left_img" id="__skype_highlight_id_left_img"><img style="width: 16px;" src="chrome://skype_ff_toolbar_win/content/famfamfam/us.gif" title="" class="skype_tb_img_flag" name="skype_tb_img_f0"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/arrow.gif" title="" class="skype_tb_img_arrow" name="skype_tb_img_a0"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"></span></span><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><span title="Call this phone number in United States of America with Skype: +17203331234" onmouseout="SkypeSetCallButtonPart(this, 0)" onmouseover="SkypeSetCallButtonPart(this, 1)" class="skype_tb_injection_right" id="__skype_highlight_id_right"><span class="skype_tb_innerText" id="__skype_highlight_id_innerText"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1"><img src="chrome://skype_ff_toolbar_win/content/space.gif" style="margin: 0px; padding: 0px; height: 1px; width: 1px;" class="skype_tb_img_space" width="1" height="1">(720)333-4828</span><span style="background-image: url(chrome://skype_ff_toolbar_win/content/cb_normal_r.gif);" class="skype_tb_injection_left_img" id="__skype_highlight_id_right_adge"><img src="chrome://skype_ff_toolbar_win/content/cb_transparent_r.gif" style="height: 11px; width: 19px;" class="skype_tb_img_adge" height="11"></span></span></span> UID: M0102001001000000004 &nbsp; Waiting for Ring... 3 seconds"

and if I hit ok it goes back to the screen I was talking before

What could be wrong
Please help

[mflorell]

What is the version and build of vicidial.php?

[john_usc]

Thanks for your replay mate!
Here is the agc/vicidial.php stats
$version = '2.0.4-148';
$build = '80519-1425';

Thanks

[mflorell]

I would suggest upgrading to 2.0.4.1rc5 to fix that bug.

[john_usc]

How do I do upgrade. Where do I get the file from.? Should I just replace the vicidial file with the newer one that you are recommending or do I need to do more.
Thanks a lot for your reply

[john_usc]

are there any installation instructions to upgrade the vicidial.php
Thanks

[john_usc]

ok I was able to install the new version but what it does is after few seconds it shows a blue screen with this

Noone is in your session: 8600051
Go Back

Call Agent Again

whats that?

Thanks

[mflorell]

Did you run the full install for the 2.0.4.1rc5 release, or did you just copy a few files?

Did you reboot after doing the install?

[john_usc]

I did a full install but problem is still there

Thanks

[mflorell]

Did your phone ring when you logged in?

[john_usc]

I made a call and my cell rang. But I can not hear anything on my head phones. Have no idea what could be wrong.
Thanks

[john_usc]

when I make a call this is what shows in CLI

== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-e3c6,2", "8600051|F") in new stack
> Channel Local/8600051@default-e3c6,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
-- Executing AGI("Local/8600051@default-e3c6,1", "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 Dial("Local/8600051@default-e3c6,1", "sip/3033331234@voicepulse-primary|55|o") in new stack
-- Called 3033331234@voicepulse-primary
-- SIP/voicepulse-primary-08b26cd8 is making progress passing it to Local/8600051@default-e3c6,1
Jan 3 00:04:15 WARNING[22191]: file.c:1045 ast_waitstream: Unexpected control subclass '14'
-- SIP/voicepulse-primary-08b26cd8 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/voicepulse-primary-08b26cd8 answered Local/8600051@default-e3c6,1
Jan 3 00:04:39 NOTICE[2328]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'

[newx22]

download heidisql software & connect to your mysql server. If you are wise enough many things can be fixed just by setting the right data in the database.

[kolucoms6]

Check yr Network.

Make sure u get call in xlite when u login.

[williamconley]

Did your phone ring when you logged in?

I believe that Matt meant Did the AGENT'S phone ring when the AGENT logged in (as opposed to a prospect phone ringing, did the agent get a call from Vicidial to complete the login process).

And if it did, did the AGENT hear "you are the only person in this conference".

And, just to be funny, did the Realtime screen for this campaign show the agent as logged in, and/or paused?

[john_usc]

Thank you for your replies. I will try to answer one by one

TO newx22:

I have phpmyadmin installed and I am pretty good with db operations but I am not sure how its going to solve the voice problem as I can make the calls and the phone even rings which I make call to. Just no voice.

TO kolucoms6:

what network problems are you referring to. Could you please give me any idea. I have no firewall running and all the ports are open. Thanks
I didn't understan what you said about xlite call in. I am using the web interface and in the leads I have my cell phone that rings when I dial

williamconley

When I login as agent, I don't hear anything. Then I click on DIAL NEXT NUMBER and it rings my cell phone but I can not hear any voice. On the top it says in gree LIVE CALL once its connected.


once the call is connect, asterisk cli shows this


== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Executing AGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log") in new stack
-- Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-f8db,1", "sip/13038003432@voicepulse-primary|55|o") in new stack
-- Called 113038003432@voicepulse-primary
-- SIP/voicepulse-primary-09e7b5c8 is making progress passing it to Local/8600051@default-f8db,1
Jan 3 13:34:22 WARNING[14873]: file.c:1045 ast_waitstream: Unexpected control subclass '14'
Jan 3 13:34:22 NOTICE[2336]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'
-- SIP/voicepulse-primary-09e7b5c8 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/voicepulse-primary-09e7b5c8 answered Local/8600051@default-f8db,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 917203334828, 2) exited non-zero on 'Local/8600051@default-f8db,1'
-- Executing DeadAGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----6") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --26-----6 completed, returning 0
-- Hungup 'Zap/pseudo-1767787704'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f8db,2'
-- Executing DeadAGI("Local/8600051@default-f8db,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== 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


Thank you all for helping me with this. I really need to get this going and now I am getting a bit nervous. Thanks again

[john_usc]

here is a complete snapshot


[root@vici ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on vici (pid = 2326)
Verbosity is at least 21
-- Remote UNIX connection
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Jan 3 13:33:43 NOTICE[14760]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-6f94,2", "8600051|K") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Jan 3 13:33:45 NOTICE[14773]: app_meetme.c:2210 admin_exec: Conference Number n ot found
-- Executing Hangup("Local/55558600051@default-6f94,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/5555860 0051@default-6f94,2'
-- Executing DeadAGI("Local/55558600051@default-6f94,2", "agi://127.0.0.1:45 77/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6--------------- completed, returning 0
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Jan 3 13:33:54 NOTICE[14791]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-f8db,2", "8600051|F") in new stack
> Channel Local/8600051@default-f8db,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Executing AGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log") in new stack
-- Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-f8db,1", "sip/13038009876@voicepulse-primary|55|o") in new stack
-- Called 13038009876@voicepulse-primary
-- SIP/voicepulse-primary-09e7b5c8 is making progress passing it to Local/8600051@default-f8db,1
Jan 3 13:34:22 WARNING[14873]: file.c:1045 ast_waitstream: Unexpected control subclass '14'
Jan 3 13:34:22 NOTICE[2336]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'
-- SIP/voicepulse-primary-09e7b5c8 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/voicepulse-primary-09e7b5c8 answered Local/8600051@default-f8db,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 913038009876, 2) exited non-zero on 'Local/8600051@default-f8db,1'
-- Executing DeadAGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----6") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --26-----6 completed, returning 0
-- Hungup 'Zap/pseudo-1767787704'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f8db,2'
-- Executing DeadAGI("Local/8600051@default-f8db,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== 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
== Refreshing DNS lookups.

[newx22]

All vicidial is build around the database, if you have problems you gotta there, most other parts of the program are static & you and me don't even know whats inside, we just use them.

check phones table, vicidial_users table, well, look at all tables for incorrect entries & you problem will be solved.