Enterprise Voip System with Asterisk and OpenSIPS


I am searching solution for build an Voip system for my company
The main points are:

  • Handle 500 concurrent calls with good voice quality
  • Record all calls
  • Thousand of users (extensions)
  • Very complex and flexible Dial Plan (many scenarios with many kinds of condition)

As my knowledge, asterisk will be unstable about more than 200 concurrent calls, so I wonder that could I use an OpenSIPS as server and 3 Asterisk Server for media purpose?
How do you think about my idea, please help me?
Thank you!

Yes you need some sort of proxy like OpenSIPS or Kamailio to handle this load.

@satish4asterisk - really? Asterisk cannot handle 500 concurent calls ?

I didn’t say Asterisk can not handle 500 concurrent calls. In fact I have handled more than 500 concurrent calls but for some other requirements. However,
(1)Call recordings is resource hungry process and I wouldn’t prefer to allow call quality deteriorate using single server . Moreover OP has mentioned about complex and flexible dialplan. And most important
(2)Do you want to put all your eggs in one basket when you are talking about 500 concurrent calls?

How install kamailio and asterisk ?

Your question is way too broad. Please describe what you have tried so for, and the specific issues that are giving you problems. Aim for a very small number of questions, each of which can be answered in less than a minute.


I assume this is a reply to me from asterisk community to my post on “AMI performance…”.

I thought my detailed query would be helpful for you to reply. but i can make it shorter as below

I am testing my asterisk 13 certified version for 120 parallel calls. I also use AMI on the same

machine to intimated call status to other computer consoles being used by agents. The calls are simulated

using Star Trinity SIP tester tool. I tested 100,000 calls of 10 sec duration each. Queue is also used.

All calls generated and connected properly by the Queue and routed to one of the free 120 extensions.

I use AMI -command MixMonitor to record the calls at the DialEnd:ANSWER event. this is required

since I need to label the recorded file name as CALLERID_EXTN_UNIQUEID_DATE_TIME format.

I also need to communicate the file path to the remote computer console for each call. Everything is working

fine and I get all 100,000 calls queued and answered.

Except, a small number of calls like 50 nos are not recorded. i,e no voice files created.

the master.csv and queue.log files are checked - all 100,000 calls are answred properly.

the messages.log is checked, the only warning is stasis task processor exceeded 500 again has come many times.

when the call rate is reduced all calls are answered and recorded properly.

pls help to resolve.

Thank you.

best regards


It was a reply to tonyklen. I hadn’t realised it was a hijacked thread with a back history.