FreeSWITCH | Communication Consolidation

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16 or 32 kilohertz and can bridge channels of different rates.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.

a Spec Sheet is available on our Wiki.

mod_fax In The Works

Submitted by anthm on Fri, 09/05/2008 - 08:38.

We've checked in a barebones mod_fax and the community is putting together an effort to get faxing support from spandsp (http://www.soft-switch.org).

Come join the fun on IRC or help be a beta tester.

Comments Posted All Is Right With The World

Submitted by anthm on Sun, 08/24/2008 - 17:04.

I am thrilled to report that my fears were unfounded. Digium Indeed has posted all the comments on their blog discussed at http://www.freeswitch.org/node/138

They just seem to have been a tiny bit lagged. Let the spirit of Open Source prevail.

Digium Blog Appears to Censor Comments

Submitted by anthm on Thu, 08/21/2008 - 15:37.

This is pretty sad. I hope they are just short-staffed.

I read this article yesterday and was surprised to see that buried in a defensive stance against sipXecs and Nortel were comments claiming my experiences with Asterisk where "rhetoric" and calling us a "Competing Open Source Project" (I am not sure what we are competing for, I thought we were on the same side, but ok)

http://blogs.digium.com/2008/08/19/asterisk-the-global-leader-in-open-source-ip-pbx-and-telephony/

Here's what he said that caught my eye:
"You should recognize that the Asterisk rhetoric from the sipXecs and FreeSwitch teams refers to Asterisk over 4 years and several versions ago when they last looked at the feature set."

This guy clearly didn't work there when I was an Asterisk developer and he has no idea the last time I looked at Asterisk because I still have had patches added to Asterisk SVN as recently as Nov 2006. I did a heck of a lot more than "look at the feature set" thank you very much.

Anyway,

I posted a comment with my feedback and almost 24 hours later it's still awaiting moderation. The sad thing was all it said was that we should all work together.

Screen Shot Showing Comment

PDF of posting with comment.

Here is my comment anyway since they won't post it:

I didn't know open source projects had to compete. They are all free aren't they? I think that's the whole problem here. Calling my list of valid issues with Asterisk rhetoric, won't make them go away. Pretending Asterisk does not suffer from any problems and only pointing out it's strengths is not the way to make it better.
Please admit that I have done more than most people are willing to do for completely FREE to try to make Asterisk better for several years. I only know enough to itemize the issues from that *long* experience as a an Asterisk developer.
I, in fact, invented the whole idea of the "function variables" that now are rampant in Asterisk 1.6 and there are *plenty* more things I could list if I wanted to. I also see plenty of ideas we have already implemented in FreeSWITCH starting to crop up in 1.6 as well. This is the nature of open source. If Digium chooses to actually cooperate with the open source telephony community there is much to be gained for all.