Plans to ramp up my setup here

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Plans to ramp up my setup here

Postby dev_4901 » Tue Sep 12, 2006 1:51 pm

Hi guys,

I am now having ALMOST trouble free running of ViciDial here. Except for a few problems like overloading of my server here, sometimes multiple calls per agent( WHICH I CAN RESOLVE NOW :) ) i do not face much problems.

Matt, there are just a few things I wish to clarify before I ramp up my existing setup from 8 seats to about 45 seats :

1) My x206 IBM server gets overloaded with 8 agents, the loadavg touches 96% with all agents logged in and dial level at 3.0.
I guess this could be because i'm running at init level 4. Could changing it to level 3 help? If yes how many agents can my present server take.

Configuration - Pentium IV 3.0GHz, 1 GB RAM , 70 GB SATA

Also this server acts as the asterisk server, DB server and the web server.

2) How many servers do I need to setup for 45 seats?
I have read the SCRATCH INSTALL GUIDE and it shows me how to use load balance across many servers, but I guess I need to know what all elements are to be installed on which servers and the settings that they would share.

Plz advice.

Thanx in advance.

Dev Singhal.
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Postby mflorell » Tue Sep 12, 2006 2:38 pm

For more info on load-balancing read the LOAD_BALANCE.txt document.

What kind of VOIP codec are you using?

Are you recording all calls?
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Postby dev_4901 » Tue Sep 12, 2006 5:03 pm

I'm using g729, as bandwidth is a constraint here. But you can suggest me some other codec whose packet size is nearly the same as g729. I know this compression takes a toll on the processor, but can't help it. I could use g723 in place of g729.

Yes, we'll be recording all calls.

Plz advice.

Dev.
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Postby mflorell » Tue Sep 12, 2006 10:16 pm

You could try GSM, it's 8k more per channel than G279 but it's much less taxing on the CPU.

Since you are recording all calls I would recommend using SCSI drives. Recording all calls will also use a lot of CPU. It helps to use the recording delay available in the 2.0 tree to lower the CPU effect on allcalls recording
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Postby dev_4901 » Wed Sep 13, 2006 11:41 pm

Hi,
I'll try using GSM codec, thanx for the advice.

Also, you have mentioned RECORDING DELAY. I'm using astguiclient_1.1.12-3, and I have not seen this option. PLz do tell me how to get the delay working, coz this would surely affect the server performance. Can I set delay as per my requirement (like 15-30 sec delay)?

The number of servers required for the setup of 48 seats too, plz......
1) With g729 or g723
2) With GSM

Plz advice and thanx in advance.

Dev.
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Postby mflorell » Thu Sep 14, 2006 2:13 am

recording delay is available only in the 2.0.1 codebase. You can use the 2.0.1b2 release.

As for the number of servers needed for 48 seats, you will need at least 3 asterisk/VICIDIAL servers and at least one database/web server.
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Postby enjay » Thu Sep 14, 2006 10:18 am

I would almost recommend 4 servers with full recording enabled.

12 users per server (dialing at as high as a level 5) recording enabled with 25 second wait time.

-enjay
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Postby dev_4901 » Thu Sep 14, 2006 7:07 pm

Greetings our new Moderators :) ,

If I use
1) 2(two) Dual Processor Xeon Servers with 2 GB RAM and SCSI HDD as dialing asterisk servers and
2) for the DB & WebServer, I take up my already running x206 IBM

Would this be good enough?

I'm using no hardware for timing.

Also do let me know if I can some how upgrade my astguiclient-1.1.12-3 version so as to use "RECORDING DELAY".
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Postby enjay » Thu Sep 14, 2006 7:23 pm

You could do a diff between the modified files of 2.0.1 and 1.1.12-3 Matt calls out what files those are towards the end of the "Hotkeys and Hangups" post (in the Support forum).

-enjay

pretty sure it can be done..
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Postby mflorell » Thu Sep 14, 2006 10:06 pm

When 2.0.1 is released it would probably be a good idea to just upgrade, and there is an UPGRADE document that takes you through the steps of upgrading from 1.1.12-3.

There are a lot of other great new features in 2.0.1 like predictive dialing and a total reworking of alot of backend processes resulting in a more efficient and easier to install package.
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Postby enjay » Thu Sep 14, 2006 10:19 pm

Agreed you are probably better off just upgrading Im just a glutton for punishment :twisted:

-enjay
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Postby dev_4901 » Mon Sep 18, 2006 2:07 pm

Hi,
Read the load balance.txt file and even saw the new release 2.0.1 and I'm itching to get my hands on the new version :D

Now I have a few queries:

1) In a load balanced setup, each of the asterisk servers are on real IPs or do they communicate on local LAN IPs??

2) Should I try APP_CONFERENCE instead of MEETME, as u have mentioned that APP_CONFERENCE is less taxing on the processors.

Plz advice.

Dev.
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Postby mflorell » Mon Sep 18, 2006 3:03 pm

The Asterisk servers need to see each other on whatever interface their "server_ip" is on since that is the IP address they will use to connect to each other.

app_conference has a few bugs yet. I only recommend trying it on lower-loaded servers, but it is rather easy to switch from one to the other, just switch around extensions.conf entries and "extensions reload" in Asterisk.
Last edited by mflorell on Mon Sep 18, 2006 4:37 pm, edited 1 time in total.
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Postby dev_4901 » Mon Sep 18, 2006 4:09 pm

Thanx Matt,

For us to enable dialing thru VoIP, do I require only one real IP, provided by the VoIP provider or do I need different real IP's for each dialing server?

I can PM you a pictorial representation of the technology we use. But till now we have used this on only one server. At the max, I separated the DB and the web server on a different machine, different from my asterisk server.

Also plz do confirm my previous query in this thread about the servers. Enjay confirmed that for 48 agents I could do with 2 dual processor servers with 2 GB RAM as asterisk server and 1 x206 IBM server with 1 GB RAM as DB + web server.

Thanx in advance.

Dev.
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Postby mflorell » Mon Sep 18, 2006 4:39 pm

If you do not have a firewall that can direct packets for SIP then you will need to place both on your real IP network.

yes, 2 servers should work fine as long as you are not using G729 for your SIP codec.
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Postby gerski » Wed Sep 20, 2006 2:18 pm

hello matt,

can you please explain to me "2 servers should work fine as long as you are not using G729 for your SIP codec"

im planning to setup 2 server asterisk/vicidial and mysql/php.. im using g729..

and also for recordings how can i do it if i wanted to send the recordings through mysql/php server?

here is my setup:

80Gig Sata - asterisk /vicidial
2 X 250 Gig php/Mysql

do you think it is a good setup or we will just swap one 250 to 80?
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Postby mflorell » Wed Sep 20, 2006 2:59 pm

G729 is very hard on the processor, and with 45 agents you will be placing lots of calls, so if you are using G729 you will need at least 3 servers for 45 agents.

As for recordigns, what do you mean by send the recordings through php/MySQL?

as for 80GB or 250GB drives, it doesn't really matter, they are SATA and if you are recording all calls, they will fail on average within one year.
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Postby gerski » Wed Sep 20, 2006 11:17 pm

what i mean is im using my first server as asterisk/vicidial, question how can i automatically send the recordings to my second server apache/mysql.. because it can accomodate up to 500Gig while my first server only have 80Gig...
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Postby gerski » Wed Sep 20, 2006 11:38 pm

question do i need to install asterisk to my 2nd server (apache/mysql)?
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Postby mflorell » Thu Sep 21, 2006 4:04 am

No need for Asterisk on the second server.

To transfer recordings you will need an FTP server(we like pure ftpd) on your second server and you will need to configure your AST_CRON_recordings_mix... script(the one in your crontab) to be able to transfer to your second server when it mixes your recordings.

You will then have to manually delete the recordings on your first server.
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Postby gerski » Thu Sep 21, 2006 11:19 am

hello matt,

so you mean on the 2nd server i will just install an FTP server, do vsftpd works ok or not?

and another thing on AST_CRON_recordings_mix on crontab, how can i put the ip of my 2nd server? (i think this must be on support section :D )
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Postby mflorell » Thu Sep 21, 2006 2:30 pm

Any FTP server should work.

You will need to edit the actual script, the FTP login information is at the top of the script. The BASIC recording mix script does not offer FTPing of audio files.
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