Freeswitch vs. Asterisk?

VOIP of the highest quality

We’ve been experimenting with VOIP in our school, primarily for internal communication.  I’ve set up both asterisk and freeswitch servers, and have been quite frustrated with the limitations of both.

Asterisk only allows one registration to be connected to each extension.  Yes, there are ways to work around this restriction (for extension 101, set up multiple extensions – 980101, 981101, 982101, and then set up a ring group 101 that rings those extensions simultaneously), but it’s an incredibly irritating workaround.

Freeswitch does allow multiple registrations on a single extension, but it has other problems.  Some of our softphones are running over WiFi and we need SRTP for these systems.  Other hardware phones don’t support SRTP, which, while not ideal, is less of an issue because they’re connected via a physical link that we have complete control over.  Unfortunately, even with Freeswitch in bridging mode, it refuses to use SRTP on the softphone link, while using no encryption on the hardware phone link.  It’s either all or nothing.  Which means, during our testing phase, we’re stuck at nothing.  Lovely.

So should I bail on Freeswitch and switch over to Asterisk?  Stick with Freeswitch and hope that I can work out some way of fixing the SRTP problem?  Or should I just give our staff tin cans attached to Cat-6 cable and tell them that’s the new VOIP system?


Monday, Nov 30, 2015

Using Asterisk’s chan_pjsip instead of chan_sip allows you to register more than once per extension.

Tuesday, Dec 1, 2015

I also suggest you look at using chan_pjsip in modern versions of Asterisk, rather than chan_sip. It’s a much more robust SIP channel driver, and does allow you have multiple registrations for a single SIP device.

Grab me on IRC (jsmith) if you need any help.

Jonathan Dieter
Tuesday, Dec 1, 2015

Thanks. I’ll take a look at that. I’m rapidly reaching the end of my tether with Freeswitch, and if chan_pjsip fixes this one quirk in Asterisk, I really don’t have any reason not to switch.

Jonathan Dieter
Tuesday, Dec 1, 2015

Thanks. I will most likely take you up on that.

Samuel Sieb
Wednesday, Dec 2, 2015

I use Asterisk for the phone system at a private school. It’s primarily hard phones right now, but I’m working on getting the teachers to use SIP on their cell phones because the cell phone coverage is terrible there. I noticed that pjsip was available now, I’ll have to look into converting to it.

Anthony Minessale II
Wednesday, Dec 16, 2015

P.S. You can indeed bridge srtp to non srtp. See the community for details.


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