vicidial.org

I am currently in the process of adding Asterisk 11 support. It is highly experimental right now.

Things I have tested so far:

Agents Login for 1 agent
Manual Dial Outbound for 1 agent
Predictive Dial Outbound for 1 agent at a 3 to 1 ratio

Things I have not tested (and not necessarily a comprehensive list):

DID routing
Ingroups
Call Transfers
Call Recordings
Three Way Transfers

Currently I am running a stress test on my development system to see if I can break it Asterisk 11. I have 35 remote agents dialing fake numbers at a 3 to 1 dial ratio. So far it has not crashed after one hour but the load seems higher than 1.8.

I also had to create a patch for Asterisk 11's chan_local.c file. During the stress test a large number of Local channels were not resolving resulting in a bunch of LRERR statused calls. I have increased the number of calls to check_bridged in chan_local.c and that seems to have fixed the issue, but now I have to make sure that is working as well.

I am looking for a few people to assist me with testing Asterisk 11. You must know how to apply a patch file and how to compile asterisk from source. You also need to know how to enable the MeetMe application in Asterisk when you are compiling it as it is off by default. I am sorry, but if you cannot do these things you will not be able to assist me.

So far no crashes, but the load on the server is quite high. A quad core with four gigs of ram is having load spikes as high as 15.

mcargile wrote:So far no crashes, but the load on the server is quite high. A quad core with four gigs of ram is having load spikes as high as 15.

What process is eating the CPU power?

I prefer to test with a virtual machine with much less power. After all, if it works there, it'll definitely survive in a physical machine ... much easier to "break" and find problems sooner rather than later

We haven't tried much in the way of optimization yet, but here is a 4-day graph of the Server Performance,



The load spikes are the most concerning part, with spikes coming every hour or so. The good news is no crashes after a week and a quarter-million calls.

right, but what's eatin' the CPU?

Code: Select all

ps -eo pcpu,pid,user,args | sort -k 1 -r | head -11

I'm not sure exactly what the output from that means, but what I can see is that "asterisk" is right at the top by far.

mflorell wrote:I'm not sure exactly what the output from that means, but what I can see is that "asterisk" is right at the top by far.

Those are the top XX CPU hogs. (head -11 gives the top 10)

So ... asterisk is your culprit and the next goal is to find out why. I'd try leaving out modules (if there are any not in use ...).

Perhaps some asterisk debug?

Also, if you quit dialing does the CPU usage drop immediately ... or does it last for a while? Perhaps AGI scripts are getting stuck with mysql processes/streams ...?

And it looks like Music On Hold playback from AGI scripts is broken in Asterisk 11. It always plays the default moh context.

Can you use an agi script to execute a dialplan entry?

Kind of a long way to go, but we've had to do it before with some ... strange "unreliable" issues with older asterisk servers using agi scripting (was the ONLY way to get the code to reliably execute). Dialplan seems to add reliable (if slow) to the equation.

Problem wasnt actually a bug in how the command works. It was a change in how 'moh reload' works in the CLI. The keep alive script was calling that after it rebuild the config file but the audio store script had not built the directory for it. Asterisk was complaining that the moh class was invalid. All subsequent calls to 'moh reload' ignored the class. Come to find out that's because Asterisk 11 now only parses a config file once after the files last write. So because the config file didnt change it didnt parse it. I added a touch command to the audio store script and it fixed the issue. Kind of a stupid idea but easy enough to work around.

Serves you right. I had a similar issue getting some vicidial scripts to write and reload asterisk. Since the Vicidial script didn't change, the "reload" wasn't issued, but I wanted it to. LOL. Had to build my own entire section for that client's Polycom Phone Generator so it would reload if the polycom file changed.

Code is now in SVN to handle Asterisk 11 and here is a vici build of it:

http://download.vicidial.com/required-a ... eta.tar.gz

You need to run 'make menuselect' and then go into applications and enable app_meetme. This is a beta build.

Hi Michael,

Any further development and stability of Vicidial in Asterisk 11? or it still a beta version? Can you please confirm?

We are building and testing OpenSuSE packages for this right now, but the tarball that we posted months ago is now considered to be a pretty stable build, since we have used it in production for months.

Hi,

I am new to Vicidial. I already have an Asterisk server 11.13.1.
I do not want to install the tarball with Asterisk.
Is the latest version of Vicidial fully compatible with it?

We don't recommend using stock Asterisk because there are several features that will not work without the patches that we include in the tarball. We strongly recommend installing Vicibox 7.0 if you want to use Vicidial with Asterisk 11.

Thanks for the quick response mflorell. I will download Vicibox 7 right now

tested asterisk 11.21.2 and im getting deadlocks under heavy load with vicidial, where there any patches done to 11.18 ? if so where can i get those patches to port them to 11.21.2

hey guys i found these 3 patches, i like to port them to asterisk 11.21.2 to do a test but before i do i like to understand the purpose of each patch to check if it works and to make sure it doesn't break anything. can you guys help me understand the purpose of these patches and what they exactly do and how best i can check them?

thank you

http://download.vicidial.com/asterisk-p ... 20.0.patch
http://download.vicidial.com/asterisk-p ... 20.0.patch
http://download.vicidial.com/asterisk-p ... 18.0.patch

matt whats the most agents you guys have put on asterisk 11 in production? we keep getting these errors "Exceptionally long voice queue length queuing to" and eventually it causes all live sip and iax2 channels to drop. any feedback would be greatly appreciated

We've experienced loads similar to those with Asterisk 1.8. There really doesn't seem to be much difference in overall performance actually. We're on our third build of Asterisk 11 in production at the moment, we've been running them in production since August. At peak loads, we are putting about 8,000 calls per hour per server through them. The latest server running Asterisk 11 that we put into production is a direct vicibox 7.01 install, and it has been running just fine for about a week.

mflorell wrote:We've experienced loads similar to those with Asterisk 1.8. There really doesn't seem to be much difference in overall performance actually. We're on our third build of Asterisk 11 in production at the moment, we've been running them in production since August. At peak loads, we are putting about 8,000 calls per hour per server through them. The latest server running Asterisk 11 that we put into production is a direct vicibox 7.01 install, and it has been running just fine for about a week.

matt where can i get the latest asterisk source you guys use to see if that makes a difference, we keep getting these errors "Exceptionally long voice queue length queuing to.." and it causes all channels to drop.

we compile from source so i cant use a package

We are on Asterisk v.11.20.0 at this point, we just used the source from Asterisk and the patches listed above, and that's the package that's included with Vicibox 7.01.

mflorell wrote:We are on Asterisk v.11.20.0 at this point, we just used the source from Asterisk and the patches listed above, and that's the package that's included with Vicibox 7.01.

matt no sangoma support?

We haven't done any testing on that yet, it's not a common feature for our clients. The 1.8 patch might actually work just fine with it.

mflorell wrote:We haven't done any testing on that yet, it's not a common feature for our clients. The 1.8 patch might actually work just fine with it.

i dont see the patch for the meetme sounds, should i be using the one for the older version of asterisk?

is it possible for you to list all patches that you apply? thank you

The meetme sounds patches are unchanged from Asterisk 1.8. So you can use the old ones.

As for Sangoma CPD/CPA/Netborder/Lyra/(whatever they want to call it this year), there is a patch for it, but we have not done wide-scale testing of it on Asterisk 11
http://download.vicidial.com/asterisk-p ... d-11.patch

Not to be a pain in the butt, for some reason my version of asterisk with these patches keeps having issues, anywhere you can possibly zip up a patched asterisk 11.20 version so I can do a deeper comparison?

I'll see if I can find it and post it.

mflorell wrote:I'll see if I can find it and post it.

Thx matt

which version of asterisk 11 is the most stable currently?

We have several stock Vicibox 7.0.3 and 7.0.4 installs. None of them crash. Many of them are under heavy load.

Thus I can't say any of them are unstable, so none would be "more" stable. Of course, this only applies to the two versions in those two releases.