dual server problem

[rudra_ach]

Hi,

we are facing this problem in dial server environment

the dialer is is daling perectly for 22 agents, but all of sudden, the call is disconenting. at the time of disconnet the asterisk is running, but in the detail view screen the only out of 22 agent only 4 agent showing online.
once the agent is loging in the dialing is going on.

please help anybody have any idea why this is happening

regards
rudra

[mflorell]

Are both servers running ntp with the time source set to the same address?

[rudra_ach]

Are both servers running ntp with the time source set to the same address?

Yes both server has ntp set in cron and to same address as below.

30 * * * * /usr/local/bin/ntpdate -u 18.145.0.30 2>/dev/null 1>&2


regards
rudra.

[asteriskguiserver]

We faced a similar problem. But the difference is we had just 10 users logged in while facing this.
The server was working fine for some 130 minutes and had problems after that. The thing i was able to discover was that the server was overloaded.
Use "w" and "top" commands and see how much load the server sucks and which process eats more of processors memory.
Try to kill that process if it had no relation with asterisk. That MAY help you out.

[caspar]

To optimize a VICIDIAL server's performance:

- Disable the “System Performance” and “Server Logs”. These log files are enabled by default and can grow to thousands of megabytes in size.

- Set “AGI Output” to NONE.

- Implement Load Balancing if you have more than one server.

- Record all calls in wav format and schedule a job to convert it to the appropriate format after hours. The conversion of wav files usually requires a lot of system resources. In other words reschedule the default VICIDIAL cron job that converts the wav files to mp3.

[mflorell]

You could also try increasing the recording delay to not record very short calls. This can help a lot if you have low-contact leads.

[asteriskguiserver]

reschedule the default VICIDIAL cron job that converts the wav files to mp3

I am a new user of VICIDIAL and am new to asterisk too.

Is there any script in the server by default(installed as given in scratch installation) which converts .wav to .mp3?
If yes, which script does that?

If there is any cronjob scheduled in server as given in scratch installation, which job/script does that?

Thanks in advance.

[mflorell]

AST_CRON_mix_recordings_MP3.pl

You need to have lame installed to use it.

[caspar]

If you have followed the instructions in SCRATCH_INSTALL.txt, you can replace this following line in your crontab:

1,6,11,16,21,26,31,36,41,46,51,56 * * * 1,2,3,4,5,6 /usr/share/astguiclient/AST_CRON_mix_recordings_BASIC.pl

with for example:

00 23 * * * /usr/share/astguiclient/AST_CRON_mix_recordings_MP3.pl

This will schedule the server to convert the wav files to mp3 every night at 23h00. Notice the format of the cron job. FIRST the minutes, THEN the hour. Do not swap them around.

[asteriskguiserver]

Any one please tell me how many agents will a server of the following configuration support with good performance...

Pentium Dual core
805 2.66GHz.
Transcend 1 GB DDR 2 RAM.
[160GB HDD]
Internet : 1 MBPS leased line for VoIP.
g729 codec installed with 20 licenses.

CentOS 4.4 server with packages installed through "yum install command".

Asterisk 1.2.14, zaptel-1.2.12, libpri-1.2.4 (as given in scratch_install)
voxfone x100p card used ( also tried ztdummy)
astguiclient : astguiclient_2.0.2b2.

Followed scratch install for crontab and rc.local

Will an upgradation in the hardware help its performance?
I know that the configuration that i have said is not much great though.

When i tested it, the servers' performance got down after i added more than 10 agents. I faced problems like late response of customers to our queries.... customers were able to hear our voice only 2-3 seconds after the agents spoke.
Sometimes the voice was not clear and was discontinuous.

[mflorell]

G729 is very resource intensive, especially if you are dialing very fast. I would suggest trying GSM just to test and see how that works for you.

[asteriskguiserver]

Thanks.
First i used "allow=all" in sip.conf. It didnt work out.
Then i used "ulaw" and "alaw" and disallowed all. Faced the same problem.
Then we changed our SIP account. Our SIP provider said that they support only g729. So i used g729.

[mflorell]

Try getting an IAX account, they usually support GSM.

[devafree]

Hi

How many g729 licenses do you have? If your service provider is disallowing all other codecs, maybe you do not have enough licenses to make 22 calls at the same time?

regards

devafree

[devafree]

hi

Sorry , i realze my post came out different than what i mean to ask. Are you using g729 for the LAN , i.e client phones also? or transcoding? becoz if agent phones are also in g729 then only 10 calls at a time can be made, the others will disconnect?

Any idea in what order asterisk will disconnect such calls? it seems random to me.

devafree

[Op3r]

when you are dialing

post the output of this command at the asterisk cli

show g729

[rajeevpn]

The scratch install has
2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

in the crontab. All the temp-info tables get reset at 01:02 AM Server time. So if you have the server set to a timezone (say IST) other than the ones you are calling (say Eastern Time), there is a very good chance that this script gets executed while the agents are dialing and of course, everybody will get kicked out.