Vicidial and internet bandwidth.

[mr.rizwan]

I have question regarding vicidial and internet bandwidth. I am using GSM codec with my SIP trunk and I have 512 Kbps internet speed for my 10 agents. Until now my agents were dialing manually every thing is working pretty much fine I have no voice issues. Now I want to use predictive dialing. Now before I will start using predictive dialing. I want to know that 512 Kbps internet speed is still enough for my 10 agents?
I am asking this because lets say if I will set my vicidial to ratio dialing at 4. and I have 10 agents logged in and they are waiting for call now dialer will dial 10x4=40 numbers at a time. Lets assume 10 numbers were answered by live human and those calls get connected with the agents. Now I want to know if dialer is using 40 channels at a time then does it mean I have 40 outbound concurrent calls on my system. If yes then do you think 512Kbps speed is will enough for me. How it works please let me know.
Thanks

[mr.rizwan]

hello please any help on this question?

[gmcust3]

I ,may , say No, Its Not enough.

You Need Minimum 1 MB of Line.

[mr.rizwan]

I ,may , say No, Its Not enough.

You Need Minimum 1 MB of Line.

Thanks for you reply. please can you tell me how did you calculate because if there are 40 calls on the dialer and you are using GSM then according to http://www.asteriskguru.com/tools/bandw ... ulator.php min speed which is required is 1145 Kibps? which less them 1MB.. please tell me how did you calculate?
Thanks

[williamconley]

your best bet would be to use iftop to show the present usage when XX agents are on the phone with clients and extrapolate your ACTUAL usage. and LOCK OUT internet usage for anything other than vicidial.

we generally estimate 100k ulaw, 75k GSM, 50k g729 to be "safe".

BUT: you don't HAVE to crank your dialer up to 4:1 dialing just because you have it in autodial mode. also: you do not want to use "adapt" (which is true predictive if you leave "available only agents" set to "N") because you do not have enough agents for that to really work. You'd need more than 20, generally I don't recommend it until at least 25. leave it on "ratio"

So if you have 10 agents, and you set your dialer to 2:1 dialing and set your MAX TRUNKS to a low number so that when your agents are all OFF the phone it does not get freaky and dial 40 lines

limit it to 20 for now, since that's more than you've ever used so far, and adjust up or down based on your call quality experience. set this under Admin->Servers. Max VICIDIAL Trunks = 20; Max Calls per Second=2 (until you have 20 agents this is probably plenty).

also consider g729 (just for communication with your SIP provider, not the agents).

[mr.rizwan]

williamconley Thanks alot for your suggestion that really helped me alot. thanks once again.
Regards
Muhammad Rizwan

[williamconley]

vicidial is a powerful tool ... but linux/asterisk/vicidial power comes with some learning

[actiondave]

also consider g729 (just for communication with your SIP provider, not the agents).

William - When you say the above, what does this mean exactly and how does it get implemented?

[randy_delgado_03]

@actiondave

William is referring to the codec(g729).

Because: g729 uses much less bandwidth than g711

But: g729 takes more processing power than g711

So: It all depends on your needs and resources.

[actiondave]

Thanks..

I was referring to him saying to implement the g729 (just for communication with your SIP provider, not the agents)

and assuming using ulaw for the agents.

How is that done and what would be the difference in bandwidth use?

[randy_delgado_03]

Thanks..

I was referring to him saying to implement the g729 (just for communication with your SIP provider, not the agents)

and assuming using ulaw for the agents.

How is that done and what would be the difference in bandwidth use?

Basically you're going to have multiple trunks in a telephony server, majority are agent trunks and the other one is/are trunk/s for your provider. William is talking about the codec config on the trunk leading to the provider. You can configure it on the SIP/IAX2 trunk but you can only use it if you already installed the codec ofcourse ...

[williamconley]

each sip account has "disallow=all" and "allow=xxxx" if you put g729 in some, ulaw in others, and both in yet others you can control which sip connections are made with which protocols. ie: phones vs carriers.

[actiondave]

makes sense now..

thanks

[telecom1777]

Yes, I would concur with that.

If you have 10 agents and dialing 4:1 and running on G711 figure at least 100k with overhead factored in so if all 10 are simultaneously on then you should be fine with 1 MB as long as no other IP applications are sharing the bandwidth.

G729 30-50k per Live Agent.