Freeswitch 1.0.0 Phoenix Released

kerryg
Posts: 6659
Member Since:
2006-05-31

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.

http://www.freeswitch.org

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



jahyde
Posts: 1996
Member Since:
2006-06-02
does your boss know about

does your boss know about this;)?

--

--my PBX is run on 2 V8's



schmooze
Posts: 233
Member Since:
2007-02-17
I have been following

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.

--

Tony Lewis
Schmooze Communications LLC



kerryg
Posts: 6659
Member Since:
2006-05-31
That would be awesome to see

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.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



schmooze
Posts: 233
Member Since:
2007-02-17
Well lets get a fund raiser

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.

--

Tony Lewis
Schmooze Communications LLC



SkykingOH
Posts: 7061
Member Since:
2007-12-17
Here is a deal for you

Sent I clearly have too much time on my hands I will make a public offer:

  1. After we quantify the cost I will make a substantial financial commitment
  2. I will commit 15-20 hours a week to PM and maintain the feature list

Scott

--

Scott

aka "Skyking"



PowerOn
Posts: 44
Member Since:
2007-10-17
Just what I need, yet

Just what I need, yet another software PBX to learn!



SkykingOH
Posts: 7061
Member Since:
2007-12-17
If this is done right

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"



PowerOn
Posts: 44
Member Since:
2007-10-17
It doesn't matter how good

It doesn't matter how good the "abstraction layer" is. Your still going to need to know how the underlying architecture works sooner or latter one way or another. That has been my experience anyways. Microsoft, Linux, etc.



kerryg
Posts: 6659
Member Since:
2006-05-31
I wouldnt concern myself

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.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



gbrook
Posts: 216
Member Since:
2006-06-06
I am really interested in this direction

I offer to provide some money and hours to test features/functionality on this new and exciting platform. I can also do some of the setup documentation for these as well.

Cheers
Garry



UncleWard
Posts: 355
Member Since:
2006-05-31
Moolah

Provided a usable and GPL2 version of FreePBX for FreeSwitch is available before Ground Hog's Day, 2009, Nerd Vittles is in for $500 and the PBX in a Flash project will contribute $500 to help fund the project.



PowerOn
Posts: 44
Member Since:
2007-10-17
Is freeswitch even ready for

Is freeswitch even ready for prime time? Prime time as in serious business environment not in someones basement.



kerryg
Posts: 6659
Member Since:
2006-05-31
This would be extrememly

This would be extrememly valuable to everyone involved and would benefit the entire FreePBX user base, so Fonality is committing $1000 to this project.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



schmooze
Posts: 233
Member Since:
2007-02-17
Wow so in a few hours we

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.

--

Tony Lewis
Schmooze Communications LLC



kerryg
Posts: 6659
Member Since:
2006-05-31
I think we need to

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.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



bubbapcguy
Posts: 3694
Member Since:
2006-06-02
Freeswitch

It is easy to install runs fine, mades IP to IP calls with great sound

the IVR and so works fine as well



lazytt
Posts: 208
Member Since:
2006-06-21
Get it done for free
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>

--

Moshe Brevda (@mbrevda), FreePBX Development Team
FreePBX tips and tricks



UncleWard
Posts: 355
Member Since:
2006-05-31
Let's keep our eye on the

Let's keep our eye on the ball this time around and not lose sight of who the real competition is. At least we can disagree on things without accusing each other of being born out of wedlock. Who's Yo Daddy!



kerryg
Posts: 6659
Member Since:
2006-05-31
There are things that are on

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.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



anthm
Posts: 5
Member Since:
2008-05-28
Sounds Like a Plan

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.



bubbapcguy
Posts: 3694
Member Since:
2006-06-02
Freeswitch

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.



jfinstrom
Posts: 1879
Member Since:
2007-03-07
I wont say I understand

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...

--



PowerOn
Posts: 44
Member Since:
2007-10-17
I had a look and one thing I

Where is the documentation? I spent some time on the wiki but couldn't find much if any configuration information. Is it all in the software package? The feature I would like to look at is SLA.



anthm
Posts: 5
Member Since:
2008-05-28
SLA

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



jahyde
Posts: 1996
Member Since:
2006-06-02
oh my, sofia has come a long

oh my, sofia has come a long way, and to think i dated her back in high school.

--

--my PBX is run on 2 V8's



jfinstrom
Posts: 1879
Member Since:
2007-03-07
Please post the bounties for

Please post the bounties for this @ http://wiki.freeswitch.org/wiki/Bounty

--



lazytt
Posts: 208
Member Since:
2006-06-21
Quote: ...even if it means
Quote:
...even if it means that code gets put into FreePBX...

Oy Vay!

--

Moshe Brevda (@mbrevda), FreePBX Development Team
FreePBX tips and tricks



kerryg
Posts: 6659
Member Since:
2006-05-31
Trolling?. I was making the

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.

--

Kerry Garrison
http://www.VoipStore.com - http://www.888VoipStore.com
Facebook: http://facebook.com/VoipStore
(888) VOIPSTORE - (888) 864-7786



teleweb
Posts: 189
Member Since:
2006-11-27
Exchange

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!



ethans
Posts: 517
Member Since:
2007-01-16
You can do Exchange 2007

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.



ADDMan
Posts: 77
Member Since:
2008-04-01
Interesting how the bounties

Interesting how the bounties vaporize when people are asked to offer it in official channels...

--

Asterisk My Anti-Drug



SkykingOH
Posts: 7061
Member Since:
2007-12-17
Quote: Interesting how the
Quote:
Interesting how the bounties vaporize when people are asked to offer it in official channels..

What is that supposed to mean? Nobody withdrew their bounty (including me).

Scott

--

Scott

aka "Skyking"



ADDMan
Posts: 77
Member Since:
2008-04-01
There is an oficcial channel

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...

--

Asterisk My Anti-Drug



SkykingOH
Posts: 7061
Member Since:
2007-12-17
Well, OK fair enough. I

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:

  1. Put together a base Scope of Work
  2. Create feature list
  3. Zero base feature list and associate bounty with features
  4. Put together a list of financial commitments to the bounty
  5. 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"



tusc
Posts: 3
Member Since:
2006-07-08
Open Source VoIP: Asterisk or FreeSwitch?

pcourtney
Posts: 1
Member Since:
2008-12-12
Freeswitch and SIP Foundry

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

www.teledesign.co.uk



pcbytc
Posts: 141
Member Since:
2007-12-04
Barracuda decided to make it (and a buck or 2)

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



SkykingOH
Posts: 7061
Member Since:
2007-12-17
Quote: even public CID, it's
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"



joshpatten
Posts: 611
Member Since:
2007-01-20
Scott: I have seen how to

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).



pcbytc
Posts: 141
Member Since:
2007-12-04
Perhaps retarded was a bit much...

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.



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