Freeswitch 1.0.0 Phoenix Released
Submitted by kerryg on Tue, 05/27/2008 - 11:53am.
If you haven't noticed, Freeswitch has released their version 1.0 version yesterday. Freeswitch is a powerful alternative to Asterisk. Unfortunately it is not compatible with Asterisk so it isn't a drop-in replacement. It is still worth looking at it from a technology point of view.
I have been following
Tue, 05/27/2008 - 12:59pm
I have been following freeswitch closely for months now and have been pounding on the FreePBX team to redesign FreePBX so it could work with numerous alternatives including Freeswitch. To do this would require a overhaul on some remaining dark sides of FreePBX but I think it would be awesome and I know Greg (Groogs) has put some serious thought into it so hopefully in the future this could become a reality.
That would be awesome to see
Tue, 05/27/2008 - 1:50pm
That would be awesome to see FreePBX support FreeSwitch. Since most of the data is contained in a mySQL database, creating an exporter to FreeSwitch isn't insanely hard. There are other parts that pull directly from config files that like voicemail passwords that would need an abstraction layer created.
Well lets get a fund raiser
Tue, 05/27/2008 - 2:09pm
Well lets get a fund raiser together to help fund this project since we all know that money talks and bull**** walks. We are willing to fund $1000.00 for this project since I feel it would be great to have another platform to use for things like large call centers and high volume auto dialing.
I'm sure Andrew would love a GUI for his trixswitch project.
Here is a deal for you
Tue, 05/27/2008 - 2:48pm
If this is done right
Tue, 05/27/2008 - 2:54pm
If this is done right Kerry's abstraction layer would make the engine transparent to the average user.
There are some compelling architecture differences that would allow for features that are very difficult in Asterisk.
How we would manage features tied to one engine and not the other in the same train is not something I have an answer to (yet).
Scott
--
Scott
aka "Skyking"
It doesn't matter how good
Tue, 05/27/2008 - 2:58pm
I wouldnt concern myself
Tue, 05/27/2008 - 3:01pm
I wouldnt concern myself about things that FreeSwitch can do that Asterisk cannot do right now. I would take the existing FreePBX feature set, which is a very complete system, and make it work on top of FreeSwitch first. FreePBX doesn't support every feature that is in Asterisk 1.4 so there is no need to add more features just for FreeSwitch yet.
Wow so in a few hours we
Tue, 05/27/2008 - 7:23pm
Wow so in a few hours we have managed to raise $3000.00. Not a bad start. Kerry we should tag team and beat up some of the card and phone manufactures for some money also since it benefits them. I think this could be a great rallying point for all of us to come together on. I am not saying freeswitch is better than asterisk or more mature it just offers some different things that asterisk can not do and it also has lots of things that it can not do that asterisk does very well its all about choices and options.
I think we need to
Tue, 05/27/2008 - 7:43pm
I think we need to understand what the expected amount of effort is going to be to make it happen, who is going to do it, and some realistic expectations of delivery dates. With that in hand it would be pretty simple for me to gain some support from some of our hardware partners.
Get it done for free
Tue, 05/27/2008 - 10:59pm
Quote:
That would be awesome to see FreePBX support FreeSwitch. Since most of the data is contained in a mySQL database, creating an exporter to FreeSwitch isn't insanely hard.
Well, considering that it "isn't insanely hard", all FreePBX has to do is to say no to Kerry and then Fonatily can build it in their forked version of FreePBX.
After all, "isn't insanely hard"!
< grin> p.s. Then, being gpl'd, we all know what FreePBX could do... < /evil gpl grin>
There are things that are on
Wed, 05/28/2008 - 6:13am
There are things that are on our roadmap to do that we are committed to doing and there are things like this and the SLA module that we would like to see but are not going to put our own resources on so it makes sense for us to contribute funds to help those projects happen to the benefit of everyone. Gosh darn it I know how hard you want to keep flaming me but our intention is to make the trixbox project better and if we can contribute to specific things that we think have value even if it means that code gets put into FreePBX for everyone to use then it is still worth it to us. We have things we want to do that do not always benefit the entire FreePBX user base and those things we will do. We feel that the SLA project and this project would benefit us so we would contribute to it, but it would also benefit the entire FreePBX user base as well. That is not a negative for us. You may want to think we are evil bastards, but we really aren't.
Sounds Like a Plan
Wed, 05/28/2008 - 7:16am
Hi,
I noticed this posting and I thought I would say hello.
I'm Anthony Minessale, lead dev on FreeSWITCH.
Having helped code most of the interfaces you probably use right now to talk to asterisk, I can assure you that you will have and easy time trying to connect to FreeSWITCH to share out the same user database, SIP accounts etc. If you need any questions answered along the way drop by and say hi. We are on irc in #freeswitch on irc.freenode.net
A few of the things you have to look forward to:
*) XML based user directory that can be delivered on demand from a CGI
*) XML based dialplan that also can be delivered on demand from a CGI
*) Multiple SIP UA on dedicated IP:PORT each able to hold multiple
external registrations.
*) Transparent voice mail integration any user who exists in the
directory has voicemail.
*) RSS/WEB access to voice mail that actually is tied to the
MWI messages.
*) XML-RPC and socket based access to the module API interface to poll
data and execute commmands.
*) Multiple registrations to the same SIP accounts with full
presence(SLA/BLF) on inbound calls.
*) ODBC and/or sqlite powered sip presence/registration database
that can be shared with a controlling application.
*) Trivial conferencing call intercept, group dial, fifo queuing/parking
all with presence.
*) GoogleTalk presence/voice/chat interop.
*) Tons of events that can be used to generate pop-ups etc.
Freeswitch
Wed, 05/28/2008 - 7:46am
Hi Anthony glad you stopped by.
Freeswitch is very nice, fast , eay to install and I have messed around with the xml files to create trunks and so on.
simple yet it works.
I did a source install on a centos 4.x load and it worked fine, I also installed into a OpenVZ centos 4.x container with very little issue.
using the down and dirty script failed twice but the boxes are on a wireless link so it could have been on my end.
One fail was in the build process, it just stopped building and hungup until reboot
The second was after the build on starting freeswitch the box would give a segment error and freeswitch would exit.
On the Openvz VPS the voice is a little fast, are you using the Hi_res timing in the 2.6x kernel???
All in all it is a nice start.
I wont say I understand
Wed, 05/28/2008 - 8:32am
I wont say I understand freeswitch but as far as ease of install I put it on a clean centos 5 box with little/no effort. Configuration is another beast, especially when your brain is stuck in Asterisk mode. I can configure an asterisk box on my blackberry going 90 down the freeway (not that I ever have or would :s ) I think FreePBX would be an asset to the Freeswitch project allowing it to attract a larger audience and gain a stronger foothold...
--
SLA
Wed, 05/28/2008 - 10:13am
We are working on the documentation but it's been 2 years of non stop coding for me so I am finally getting enough breathing room to work on docs.
http://fisheye.freeswitch.org/browse/FreeSWITCH/src
See the box in the upper left corner, 0 to 250k lines of code in 2 years and most of that red line is me so it's hard to write docs all by myself. That's why our community has been a big help with all of that wiki stuff you saw.
SLA is enabled in the sip profile
http://wiki.freeswitch.org/wiki/Sofia
http://wiki.freeswitch.org/wiki/Sofia#Multiple_Registrations
the default sip ua on a new install will be at:
/usr/local/freeswitch/conf/sip_profiles/internal.xml
<param name="multiple-registrations" value="true"/>
by adding that to the sip profile (the instance of a UA) it allows multiple registrations to the same account.
The other important config option is manage-presence
<param name="manage-presence" value="true"/>
This is enabled by default on the "internal" profile already.
With these 2 params most SLA/BLF functionality will work on devices that use dialog-info for presence, I am actually fixing up a patch to make the polycoms work this afternoon.
We are at the mercy of our community to provide documentation so if anyone has any questions you should pose them to the mailing list and/or our IRC channel #freeswitch on irc.freenode.net you will find that they add things to the wiki as fast as possible. In fact, we have a conference call wiki meeting today in about an hour at sip:888@conference.freeswitch.org
btw: digg it =D
http://digg.com/software/FreeSwitch_1_0_Released
Please post the bounties for
Thu, 05/29/2008 - 9:46am
Trolling?. I was making the
Thu, 05/29/2008 - 3:25pm
Trolling?. I was making the point that we are willing to help fund this even though the benefit is not 100% to our own benefit which is something people accuse us of a lot. We have put up bounties for other projects that are being designed for FreePBX as well. This is just one that I feel strongly about supporting as it is a significant coding project and we would rather contribute to its overall success than doing it ourselves for purely our own gain. Sorry if you took that as anything different.
Exchange
Thu, 05/29/2008 - 6:08pm
anthm, since Freeswitch supports SIP/TCP, does it integrate nicely with Exchange 2007 UM?
Support and interoperability with Exchange 2007 Voice Access has been one of THE most frequently requested features here on the Trixbox forums, so if your product does it, that's another major reason to consider it!
You can do Exchange 2007
Thu, 05/29/2008 - 7:20pm
You can do Exchange 2007 integration now with Asterisk using a sipX gateway for SIP over TCP, but it's a nightmare from what I can gather. We've had Exchange 2007 for a few months now, but I haven't had the time to integrate it yet.
The biggest question is does FreeSwitch support uaCSTA for OCS 2007 integration? Talks about uaCSTA Asterisk Manager Interface integration in Asterisk seem to have stalled, and the commercial solution I've seen is $500/server, which makes the solution cost prohibitive for mainstream integration.
There is an oficcial channel
Thu, 05/29/2008 - 9:20pm
There is an oficcial channel for such bounties posted above. These channels get freeswitch developers involved which is important in such a task. I just noticed no one has put their offers up where they count... I am guessing anthony only popped in because he was directed here and it is unlikely any Freeswitch or Asterisk developers are actively monitoring in the hope this thread would happen...
Well, OK fair enough. I
Thu, 05/29/2008 - 9:27pm
Well, OK fair enough. I said I would help project manage this so let's get the ball rolling.
Here is what we need to do:
- Put together a base Scope of Work
- Create feature list
- Zero base feature list and associate bounty with features
- Put together a list of financial commitments to the bounty
- Submit the bounty as a unified group
Everybody that spoke up and offered money are "bankable" in my book so let's get the ball rolling.
I will post a draft scope of work over the weekend and we can start to hash this out.
Scott
--
Scott
aka "Skyking"
Open Source VoIP: Asterisk or FreeSwitch?
Tue, 07/22/2008 - 9:14pm
Freeswitch and SIP Foundry
Fri, 12/12/2008 - 11:56am
the SipX stuff looks to be an interesting option as they will be releasing ver 4 in March 2009
http://sipx-wiki.calivia.com/index.php/SipX_Roadmap
Barracuda decided to make it (and a buck or 2)
Sun, 10/25/2009 - 4:48am
Wow, thread just up and died?
Well, I just bought a CudaTel switch, based on FreeSwitch, and although they haven't brought out everything to the user interface, HD Voice, SLA works (no PARK yet) and, oddly enough, when you make a call, it does a CID lookup and displays that on the calling phone, even public CID, it's cool and weird at the same time? I have no clue how they do that?
I was trying to roll to sipXecs from Trixbox originally (Mainly for HD Voice & Exchange 2007 integration), but I could not get a call to transfer that came in via a trunk (to my Cisco or the existing Trixbox) and I was tired of the damned excuses "oh, trixbox is not sip compliant" yeah, well my Cisco router is, you sipX guys have this retarded purist approach to how SIP should be handled by the world, guess what, not many are compliant and YOU need to compensate or be ignored (by people like me, who need things to just WORK) CudaTel WORKS, has 24 hour support, and it's really cheap for what it does. (But does NOT support Exch2007 integration)
I will continue to much around with FreeSwitch, It's the BOMB!
Bye Bye sipX, (And I keep trix around for the custom crap I don't know how to do on FreeSwitch)
TimC
Quote: even public CID, it's
Sun, 10/25/2009 - 10:14am
Quote:
even public CID, it's cool and weird at the same time? I have no clue how they do that?
They have a predial hook xml event like the Aastra and Polycom. The Aastra scripts include the extension lookup by default. You could expand that to outgoing calls.
Page 2-36 of this document describes the interface:
http://www.polycom.com/global/documents/support/setup_maintenance...
--
Scott
aka "Skyking"
Scott: I have seen how to
Sun, 10/25/2009 - 1:54pm
Scott: I have seen how to send out the information to a script on Polycom phones, but not how to receive it back and display on the screen like the Aastra's do. Any info on how this could be accomplished?
pcbytc: guess you never got it figured out? That's too bad, however their "retarded" purist approach is a compilation of RFC standards and, at this point, one of the few things keeping the SIP standards "standard". There are WAY too many deviations and non-standard half-baked implementations of SIP floating around right now that, in my opinion, are hurting SIP as a protocol. Many of the developers of sipX sit on the IETF board so it is their responsibility to keep the SIP standard pure, and what's really ironic is that you praise FreeSWITCH to no end, yet sipX and FreeSWITCH are collaborating their platforms together to create a unique platform that can do some nifty things, all within SIP standards.
The blame for the failure of your sipX installation is not to be placed on any party except those that do not follow the SIP standard. From what I understood you were trunking your Cisco router THROUGH trixbox to sipX because of some feature that you needed. This isn't supported due to the fact that Asterisk has a watered down SIP implementation that is incomplete. I once had the same situation happen to me (trying to trunk through Asterisk) and I had to switch to callweaver to get decent SIP compatibility because it just doesn't exist in Asterisk 1.4 (haven't tried with 1.6 yet).
That being said, FreeSWITCH is an excellent platform and Cudatel will probably take off like a rocket if it's success is anything like Barracuda's spam firewall products (I administer one). FreePBX v3 will initially release for FreeSWITCH before Asterisk (from what I hear).
Perhaps retarded was a bit much...
Mon, 10/26/2009 - 10:49am
I guess the reason for my frustration Josh is I just haven't had that much trouble with any sip product before, and as far as the Cisco goes, I tried it direct to SIPX with a few extentions on another dial-peer, had similar trouble. Thanks for all your help though.
It just seems like the "purist" approach makes it all that much harder, I applaud the collaboration between freeswitch & sipX, hell, if sipX replaced the entire trunk portion of it's product with the engine out of FreeSwitch I would run back to it without hesitation, just for the endpoint server redundancy! SipX desperatly needs realtime call diagnostics, I hate not having that.
I am a true LinIdiot, I know enough to hack out some config files and make my phones play dixie if I want them to, but I can't compile anything to save my life. I am a Windows guy and WinSCP is my friend.
So, Until somebody cooks-up a boot ISO to install FreeSwitch on Centos (or whatever Linux) with a FreePBX front-end that will just work "out of the box", I'll buy CudaTel.
Member Since:
2006-05-31