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Softswitch opensource:Kamailio

Firstly, I tried Kamailio. This particular product is a derivative of what was originally SIP Express Router. This product, it is claimed, handles hundreds and hundreds of call setups per second, so it is very suitable for use in large telcos. Out of the three packages, it is probably the closest thing to what is known as a softswitch in telecoms terms, and is also the closest thing to how the SIP protocol was envisaged to work when it was created. Unlike the other two products, what Kamailio is best at is setting up, tearing down and routing calls. Which is great, but unfortunately people expect “features” in their PBXes these days, such as voicemail or conferencing and so on! With a softswitch, this is typically done by handing off to “feature” servers (which, ironically, are quite often run by PBXes!)
What really put the nail in the coffin for me as far as Kamailio was concerned, was how complex the configuration file was. It seemed necessary to understand the SIP protocol at a very low level to even begin to understand how to use it. Whilst I have a good book on this subject, I felt it was massive overkill to need to configure a softswitch at that low a level for what is basically a small home PBX. A pity, because I think it is probably very good at what it does, but just wasn’t for me.



  • secure peer-to-peer communication
  • person to person or group voice calls
  • person to person video calls
  • screen sharing
  • instant messaging
  • presence status
  • call detail records

So, FreeSWITCH wins, but what’s happening this weekend?

So, what now? This weekend marks the official switchover to FreeSWITCH. Currently my “live numbers” are not on any PBX at all, going straight to the phone so that I can guarantee the calls can get through whilst the new server was being set up. I have a couple of test numbers on the fs server from two different providers, so that I can test compatibility before I move all the numbers over.

The configuration isn’t finished yet, but most of it is done. Voicemail is working, and the conferencing server is up, and I can route calls inbound and outbound to and from the PSTN. Still left to do are conversions of a couple of old Asterisk scripts, plus implementation of ACR (anonymous call rejection) and other assorted bits and pieces like implementation of feature codes, but none of those will stop the incoming calls ringing the phones if they’re not in place.

So, if you can’t get through on the this weekend, that’ll be why … although I don’t expect the job will be too arduous, like most things of this nature, it’s fine if it doesn’t go wrong. I expect I’ll be blogging again after the event to let you know how well it didn’t work


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