Problem in predictive dialing,

[aali_baba]

HI,

i am new to VICIDIAL,

i am using vicidial now 1.3 with SIP Trunk,

problem i am facing is that after some times my predictive calls gets stop,
i see in campaign page " No live calls waiting, and so no calls gets transfer to agents,

after this restarting server this problem gets resolve for uncertain time , some times for 45 minutes and some times for 5 minutes calls dial through and then get stop.

i have tried

/usr/share/astguiclient/

true AST_reset_mysql_vars.pl

but it didn't work for me .

my server time is sync with npdt and so of my client end machine times.

please help me to resolve this issue.

Regards

Aali

[aali_baba]

HI,

I always think as ,

Answer can be stupid but Questions can't,

so even if you all feel my question is stupid guide or reply me please

Regards

Aali

[Michael_N]

Does manual dialing work?

Have you talked with your sip-provider about this?

[aali_baba]

yeah manual dialing is working fine

their is no issue with my sip provider as i am using same on other locations as well,

i do not see any activity on vici*CLI> command means basicaly dialer stop dialing,

[gardo]

Any other messages or errors when the server stopped dialing? Do you have enough leads loaded? What's the load average on the system?

[aali_baba]

Hi,
thanks for reply,

Load evg is in between 1 to 2.5, i use to have enough leads,

i think this is happening because of some sort of lead problems ,,,,,,
i've check lead sets completely and find fine but still i feel its something related to leads or database not the dialer,

i have 2 different campaigns running on this dialer both have abut same kind of leads ,,,,,,, but one is giving problem while other campaign is going fine, i am really stuck now ,

do you think this issue can be cause from Agent PC ? because dialer is located on one place while both campaigns are running on different locations,

please guide me i am really stuck and so stupid thoughts are coming in my mind

Regards

Aali

[gardo]

I can't really say. But that might be a factor. What are your server specs? How many agents dialing? What's the bandwidth going to the server? Asterisk CLI when this happens or before. Screenshot of realtime monitor page and agent interface.

[aali_baba]

Hi,

thanks for reply once again

what i done is i check every lead one by one in a lead set and upload,
and i sync all agents system and server from pool.ntp.org for time sync,
and my Saturday calling went smooth without any issue.


but this problem some times occurs some times after 4,5 hours and i have 4 hours shift on Saturday so i have to observer it for at least one more day and then i will update you,

Asterisk CLI gets stop ( no activity )when this problem occurs.

i have 2 MB dedicated link for this server directly terminated on server NIC,

server config is 3.2 Xeon dule prccessor with 4 GB RAM and RAID 0,

MAX 14 Agents use to log in in current scenario.

Regards,

Aali

[neoman]

Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-22aa,2", "8600051|F") in new stack
> Channel Local/8600051@default-22aa,1 was answered.
== Starting Local/8600051@default-22aa,1 at default,9397736031,1 failed so falling back to exten 's'
== Starting Local/8600051@default-22aa,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-22aa,1
-- Executing Playback("Local/8600051@default-22aa,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
May 25 08:41:57 WARNING[14677]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/58600051@default-74ef,2", "8600051|Fmq") in new stack
> Channel Local/58600051@default-74ef,1 was answered.
-- Executing Answer("Local/58600051@default-74ef,1", "") in new stack
-- Executing Monitor("Local/58600051@default-74ef,1", "wav|20100525-181155_97736031") in new stack
-- Executing Wait("Local/58600051@default-74ef,1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

My Dial Planis
exten => _961NXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _961NXXXXXXXX,2,Dial(SIP/${EXTEN:1}@SIP-outbound,55,o)
exten => _961NXXXXXXXX,3,Hangup
for any type pf other number
exten => _961.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _961.,2,Dial(SIP/${EXTEN:1}@SIP-outbound,55,o)
exten => _961.,3,Hangup

not able get call
Xlite Settings
it works othe plan for Us and Uk but not for Austrelia
Please Help M
SIP trunk used for It



VicidialNow-
Astrestik Version-1.2.30.2
VERSION: 2.0.5-206 BUILD: 90525-1014

[aali_baba]

Hi/1

i am still facing same problem,

can any body help me please !

Regards,

Ali

[aali_baba]

Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-22aa,2", "8600051|F") in new stack
> Channel Local/8600051@default-22aa,1 was answered.
== Starting Local/8600051@default-22aa,1 at default,9397736031,1 failed so falling back to exten 's'
== Starting Local/8600051@default-22aa,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-22aa,1
-- Executing Playback("Local/8600051@default-22aa,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
May 25 08:41:57 WARNING[14677]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/58600051@default-74ef,2", "8600051|Fmq") in new stack
> Channel Local/58600051@default-74ef,1 was answered.
-- Executing Answer("Local/58600051@default-74ef,1", "") in new stack
-- Executing Monitor("Local/58600051@default-74ef,1", "wav|20100525-181155_97736031") in new stack
-- Executing Wait("Local/58600051@default-74ef,1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

My Dial Planis
exten => _961NXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _961NXXXXXXXX,2,Dial(SIP/${EXTEN:1}@SIP-outbound,55,o)
exten => _961NXXXXXXXX,3,Hangup
for any type pf other number
exten => _961.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _961.,2,Dial(SIP/${EXTEN:1}@SIP-outbound,55,o)
exten => _961.,3,Hangup

not able get call
Xlite Settings
it works othe plan for Us and Uk but not for Austrelia
Please Help M
SIP trunk used for It



VicidialNow-
Astrestik Version-1.2.30.2
VERSION: 2.0.5-206 BUILD: 90525-1014

what prefix you are using for US and UK dial plans?