Vicidial/Asterisk calls dropping

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taonga

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We are a small call center running Vicidial/asterisk (Opensuse 12), Of late the agents keep complaining about being kicked out of vicidial and calls dropping. the guys managing the server and the isp are blaming each other.I would like to hear from someone else whose has some knowledge on vicidial/asterisk.

The ISP says is probably the resources of the server that are inadequate but the server guys say we need to up the the diginet line.

We have a 512kb diginet line.

How many concurrent call can you make on this line

Server
4x Intel(R) Xeon(R) CPU E5603 @ 1.60GHz, 1596 MHz
16GB RAM
2 x 1TB HDD

Lastly Please explain the difference between active channels and active calls
 
Sup!

There are two channels per call once there is two-way media set up.
512K Diginet should get you about 12 concurrent calls. This is asymmetrical for power dialling where there will be more outgoing bandwidth for call setups than actual concurrent calls.
Your server hardware looks OK but it depends on the database load for those hard drives. How many agents are on it?

Lastly, check your asterisk messages log for "Peer" to see if your SIP provider or phones are dropping.

Vin.
 
there are almost 20 agents, the problem starts when there are about 16 agents at the same time. we're thinking of getting a 5Mb fibre line as the cost is almost the same as the 512kb line. will that help?
 
I think you need to get clarity on agents getting kicked out of vicidial and calls dropping. These can be very different things. Agents getting kicked out often has to do with time issues either on the server or stamps in the db. Accurate clock is also very important in Vicidial. You can use pseudo clock, but we have seen issues with it. Have you got a telephony card or a VoiceTime? There is also a difference between a call dropping and not being able to make a call. First generally points to a connectivity problem the second to concurrent call capacity. Call drops could be at various points, Agent, your LAN, server, SIP provider issues.

Vicidial is primarily aimed at outbound dialing. Are you dialing manually or using some of the automation features to queue up leads? This will have an impact on required concurrent calls you need to be able to make. You have not mentioned how many agents you have or how many concurrent calls you are paying your SIP provider for. I assume the diginet line is dedicated for voice only?

As mentioned by vince0 Asterisk creates a channel per leg of the call and then bridges them, so that's one channel to the agent and another over the SIP provider. For simplicity we rather refer to concurrent calls when speaking to the SIP provider so it removes the ambiguity. Vicidial behaves differently to a normal asterisk call in that it uses conferences to join parties together (which is why clock is important). Agents sit and wait in the conference, when Vicidial gets a lead to answer it adds that person to the conference. In auto-dialing mode it will then start calling the next lead to line them up for an agent.

The amount of concurrent calls you can carry on the line will be determined by the codec you use. G711 needs 64k, G729 only 8k. Any respectable SIP provider should offer G729. You would however need to enable G729 on your server. There is an unlicensed version available, but if you are running this in production then you should purchase your license from Digium.

On the face of it your server specs look fine, depending on how many agents you have and what else you are running on it.

We have extensive experience with Vicidial, feel free to drop me a PM if you want to engage in formal support.

Edit: I was still typing this up when you posted the agent count. If all that server is doing in Vicidial for 20 agents, then there shouldn't be an issue with hardware.
 
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512K Diginet definitely won't cut it for 20 agents, better upgrade.
 
512k will do 20 calls using g729. The calc you provided does look like it includes some overhead because 8x20 only gets you to 160k

g729calc.jpg

In any event it should be easy for the ISP to provide graphs showing that the link is saturated. The SIP provider should also be able to provide stats on concurrent calls.
 
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I stand corrected. Best thing to do is get a usage graph and see for sure.
 
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This is what i got when I ashed for a usage graph

XdAp69.jpg
 
So looks like 15 concurrent calls if you look at "Total Average Concurrency"?

Have you confirmed the codec?
 
the major problem is call just drop in the middle of a conversation, they also say that sometimes say that they randomly get logged of vicidial and have to login again
 
i also got this info from the server guys, what does it mean

Every 2.0s: asterisk -vvvvvrx 'core show channels' | egrep "(call|channel)" Mon Sep 21 12:43:46 2015

49 active channels
28 active calls
 
taonga said:
the major problem is call just drop in the middle of a conversation, they also say that sometimes say that they randomly get logged of vicidial and have to login again
Click to expand...

Then that's a separate problem to the concurrent calls you can run on your diginet line.
 
taonga said:
i also got this info from the server guys, what does it mean

Every 2.0s: asterisk -vvvvvrx 'core show channels' | egrep "(call|channel)" Mon Sep 21 12:43:46 2015

49 active channels
28 active calls
Click to expand...

Generally speaking 28 active calls is the same as concurrent calls, but because you are running vicidial you need to take conferences into account. It can also include internal calls. So you either need to filter that down to only calls out over the sip trunk or compare that to "sip show channels". "core show channels verbose" is also useful for finding hung calls (shows Duration) and seeing more detail about the call.
 
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