vicidial.org

the agent is logged in.. you can see it on the asterisk cli... i get the message that im the only person in the conference and then i clicked resume and nothing happens and on the CLI nothing happens either just the ==Manager 'sendcron' logged on and off

Do you have dialable leads in a list, in the campaign (and active)?

Do you have calls in your hopper?

yes i do.... autodial i have it on 1.... 10 leads in hopper and 94 lead in this lists
1 active and 0 inactives..

What are the result of running this in your linux command-line:
screen -r

There are several suitable screens on:
2952.ASTVDremote (Detached)
2934.ASTlisten (Detached)
2945.ASTVDauto (Detached)
2623.asterisk (Detached)
2940.ASTsend (Detached)
2953.ASTupdate (Detached)

attach to this screen and post some output when you have and agent logged in and RESUMEd:
screen -x ASTVDauto

(to detach, type "Ctrl a" then "d")

loop counter: |5759981|
TIME DEBUG: -5|-5|0| GMT: 16:39
2006-08-02 11:39:54|LIVE AGENTS LOGGED IN: 1|
2006-08-02 11:39:54|CALLCAMP 127.0.0.1: agents: 1 dial_level: 1|
2006-08-02 11:39:54|CALLCAMP 127.0.0.1: Calls to place: 0 (1 - 0) MVT override: |
2006-08-02 11:39:54|CALLCAMP 127.0.0.1: CALLING|
TIME DEBUG: -5|-5|0| GMT: 16:39
2006-08-02 11:39:54|| lagged call vla agent PAUSED 0|20060802113924|20060802113944|20060802113954||
2006-08-02 11:39:54|| lagged call vac agent DELETED 0|2006-08-02 11:37:54

this is when i typed in screen -x ASTVDauto

Is this ona single server? If not and the database/apache server are seperate from the vicidial/asterisk server you need to ensure that they both have NTP synced..

this is a single server with apache, mysql and vicidial on the same computer...

do you have any records in your vicidial_auto_calls table?

do you have the two entries in your dialplan for the "h" exten?

yes i have
exten => h,1,DeadAGI(call_log.agi,${EXTEN})
exten => h,2,DeadAGI(VD_hangup.agi,${EXTEN})

if i go to phpMyAdmin.... and clicked on asterisk database.. then i go to vicidial_auto_calls nothing is there...like i try go to to browse and i can't.. it just show me the structure.

do u have the dial option o

dial option o?????? where can i see that?

exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN},55,o)

i have

exten => _91XXXXXXXXXX,1, Dial(SIP/${EXTEN:1}@telasip,60,tTo)
exten => _91XXXXXXXXXX,2, Hangup();

have u try this one? .. i cant see that u have a call_log.agi

exten => _91XXXXXXXXXX,1, AGI(call_log.agi,${EXTEN})
exten => _91XXXXXXXXXX,2, Dial(SIP/${EXTEN:1}@telasip,60,tTo)
exten => _91XXXXXXXXXX,3, Hangup();

ok.. once i do that change do i have to reload asterisk from the cli???
another thing.. when i logged in with one of my agents it comes up on the cli after it shows that is playing the message of "you are currently the only person in this conference" and creates the meetme 86000052 it comes up with a notice here
NOTICE[9693]: rtp.c:575 ast_rtp_read: Unknown RTP codec 126 received.

yez, i would recommend to always reload asterisk when making any changes in the config file.

wat codec are you using?
are you using a softphone or hardphone?
try to explicitly allow the codec you are using in sip.conf

im using a softphone Xlite...well actually the EYEBEAM... also my
[general]
port=5060
bindaddr=0.0.0.0
context=default
tos=lowdelay
disallow=all
allow=ulaw

i fixed it...
it was a problem with the time on the server....
the agent had -5 on the GMT
and i changed it to 0 and is working fine...

Do you have the phone_code field populated in the vicidial_list table?

Are you running the adjust gmt script in the cron?