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is there a limit on number of conferences??

PostPosted: Sun Apr 26, 2009 2:28 pm
by arpit.modi
Hello all,

i have setup vicidial version 2.0.4 works great with my requirements, but when number of agent logins are increasing, i mean 25-30 agents are logged in and they are dialing from different campaigns, sometimes they can hear each other's voices.


i think conferences are getting stucked, but when i see in asterisk it shows me proper conferences.

but even though restarting asterisk its again working fine....

i am using asterisk 1.4.21

so i am thinking to increasing number of vicidial conferences,

will it help me to solve my problem?

or any ideas to solve this issue.............


please help me


thanks in advance.

PostPosted: Sun Apr 26, 2009 4:17 pm
by williamconley
I would not expect increasing the number of conferences to help, actually.

I suspect server load, but at this point it is purely a guess because I really have no idea what your configuration is like.

Can you read the "stickies" for this forum, then read the manager's manual (in case you haven't, it is required reading).

Then please post your configuration (software versions for Vicidial, asterisk, zaptel, etc, along with hardware and OS in use).

It would also be helpful to know if you can reproduce this occurrence and what is required to do so (if it happens reliably when you go over 20 agents, for instance). And, of course, if that is the case it would be good to know your server load as well. (May come in handy regardless.)

And for those who do it (and you KNOW who you are!) "Not very much" is NOT a statement of "server load". LOL. It is easiest to get this way (but there are several methods):
Code: Select all
root@vici:~# uptime
 14:23:11 up  7:37,  2 users,  load average: 0.08, 0.08, 0.12

PostPosted: Sun Apr 26, 2009 5:11 pm
by Op3r
Either they dont log out properly or you are maxing out your loads.

You can also get your load average by typing

root@kamote# w

or

root@kamote# loadavg

or

root@kamote# top
(look at the load portion)

using 3 different servers

PostPosted: Mon Apr 27, 2009 2:39 am
by arpit.modi
hi thanks for the reply,


i am using 3 different severs,

1 for web UI,
configuration is

Intel(R) Pentium(R) Dual CPU E2180 @ 2.00GHz
1 gb ram
136 gb harddisk
OS : CentOS release 5.2 (Final)

1 for database,
configuration is

Intel(R) Xeon(R) CPU X3220 @ 2.40GHz
4GB ram
130 GB harddisk
OS : CentOS release 5.2 (Final)



1 for asterisk,
configuration is

Intel(R) Xeon(R) CPU X3220 @ 2.40GHz
4GB ram
130 GB harddisk
OS : CentOS release 5.2 (Final)


So, i think there is not much server load as we have distributed the load.

but may be this is because of whenever i logout agent, the channel is not being hangup at that time....

now can you please suggest me some solution?

Thanks, again.

update with asterisk log

PostPosted: Mon Apr 27, 2009 3:02 am
by arpit.modi
hey all one more thing....


i use IAX like zoiper phones.


and when i logout agent it shows following in asterisk,



== Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/1001-13127'
-- Executing [h@default:1] DeadAGI("IAX2/1001-13127", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing [h@default:2] DeadAGI("IAX2/1001-13127", "VD_hangup.agi|PRI-----DEBUG-----0---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/1001-13127'


localhost*CLI> show channels
Channel Location State Application(Data)
SIP/vicidial-082938b 8600051@default:1 Up MeetMe(8600051)
1 active channel
1 active call>


that seems, the conference is not being released at that time..............
hope anyone will get some ideas......


thanks again

PostPosted: Mon Apr 27, 2009 9:42 am
by Op3r
do you have the exten =h entries in your context?

PostPosted: Tue Apr 28, 2009 2:56 am
by arpit.modi
there are entries of just dial and hangup


exten => _X.,1,AGI(call_log.agi);
exten => _X.,2,Dial(SIP/vicidial/${EXTEN},,tTo)
exten => _X.,3,Hangup

PostPosted: Tue Apr 28, 2009 3:28 am
by Op3r
what are you using?

That entries was like 2 years ago. We move to fastagi now :(

PostPosted: Tue Apr 28, 2009 10:52 am
by williamconley
You should probably go back to using the extensions.conf file supplied with vicidial and only modify the [globals] section ... you will probably find that it works better that way.

Any changes to the dial plan will result in a very nice education (about why what you changed had unexpected results) ...

But if you want to skip the education and get the system acting appropriately: limit changes to duplicating existing dial plans (like duplicating dial 9 service to make dial 8 service) from the supplied extensions.conf is cool, making up your own is not.

If the only thing you modify is the [globals] variables definitions ... it works much smoother.

Or upgrade to the latest version (vicibox or vicidialnow ...)