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The FreeSWITCH bunch have a useful saying: "Don't glue the Lego pieces together".

A modular system is best enjoyed as a modular system. This is one of the most powerful things about unix systems, you can pipe the output of cat to the input of sed and feed that to a text file for editing by vi.

I don't have a dog in this systemd fight, and I agree with another poster who thinks this is much drama over not much. It is a sea change in how to think about things, but I certainly do not miss hunting down a missing sysvinit script and slogging through it to see why it doesn't work because of one silly line in it. Building a unit file is child's play by comparison. As long as the damned thing works I'm fine with it.

We seen this coming long ago, we did a lot of work to make sure we were IPv6 Ready, Check it out on http://www.freeswitch.org/

In fact, to support your comments about Javascript, in the FreeSWITCH project, we have a VoIP softswitch that can directly interact with Javascript using mod_v8 (used to be spidermonkey), and can also interact with lua, perl, and other languages - scripted, compiled, managed, etc. in a similar fashion.

Especially for multimedia manipulation. Our project FreeSWITCH http://freeswitch.org/ needed most of the updates in jessie to be able to run properly. All the libraries like libavcodec, libavformat and vlc etc. it's harder than it looks to swap out libraries because you need harmony among all the software it supports. Sometimes changing one library can cause a lot of issues that are not always immediately visible. New releases, even if not exciting on the outside, often have a lot going on behind the scenes.

In anticipation of this transition to hangouts, We have added WebRTC support to FreeSWITCH, our Open Source Telephony and Voice application framework.
We have had support for Jingle for many years allowing communication with google voice and I suspect we will be able to use our new WebRTC functionality to connect Google hangouts to any voice applications you can make using FreeSWITCH http://www.freeswitch.org/

We are featuring WebRTC applications in August at our ClueCon conference http://www.cluecon.com/

I felt like chiming in here a bit, as I think I understand what serviscope_minor is getting at. I too have contributed to several projects - whether it be documentation, fixes to code, or enhancements (most recently I added some functionality to FreeSWITCH's mod_curl. I've been down this path before, but I'm also experienced enough in life and professionally to see things for what they really are. Most of the time, you'll be told that your implementation is wrong and needs to be changed, or can it be done a different way, the idea is not good, or they don't like your edits. Now there's two possible responses that you, the contributor, can give:

1) You can cry because somebody was blunt and/or curt with you about your contributions, and take your ball and go home.

OR

2) You can avoid taking the criticism personally. From my experience, there are almost always very good reasons for this criticism. Just because you are providing volunteer work doesn't mean that the project in question is required to accept your contribution. You are either just trying to contribute what they don't need or don't want, or you are contributing something they do want but you need to do it in a different way so it fits better with the vision of the project.

BasilBrush - This is why he is asking for TWiTfan to provide the context. The context is all important - did TWiTfan choose 1 or 2 above? Do we agree that his choice to do one or the other was a good thing or a case of overly thin skin and too easily bruised of an ego? This isn't about 80's frats. This is simply about the cruel nature of the world: the vast majority of the time people only care about what you can provide to them, and if you can't provide anything, then scram.

CudaTel (of Barracuda spam firewall fame) appliances are built on top of Freeswitch, an open-source PBX that I've found scales much better than Asterisk. The hardware is sized by number of concurrent calls. If you don't know how many concurrent calls you handle, the accepted convention is to take the number of phones you have and divide by 6 (or 4 if you want to be very conservative).

It's not practical for VoIP providers to offer encryption most of the time, because their connections to the real POTS/PSTN is still just regular, wiretappable PRI/T1s at some point along the line. They have to interconnect with the real phone network at some point to be useful, and all calls therefore are still tappable.

However, you could just use Zphone with ZRTP (or run your own PBX using FreeSWITCH to accomplish what you are looking for from a VoIP provider).

As a current Asterisk 1.6 user, I can attest that it is a piece of junk. It's monolithic, buggy, poorly documented and unwieldy to install from source (witness the number of ISO based all in one installation solutions).

I'm in the process of reading up on FreeSwitch with a view to shifting to it.

Have a read of: http://www.freeswitch.org/node/117

A good PBX does not allow the dial plan to dynamically change or jump to different extensions then what's intended. In fact, it wouldn't jump or "Goto" extensions at all. Everything should be defined in a formal language with an actual grammar and stored in read-only memory.

There are already projects which have learned (the hard way) from Asterisks mistakes, such as Afelio and FreeSWITCH. I would've started with one of those, because they make the mistakes of Asterisk a lot more obvious.

I have been working on the open source softswitch FreeSWITCH http://www.freeswitch.org/ for almost 6 years now.
During that time I have seen SIP continuously evolve to try to cover its own shortcomings which all stemmed from the simple concept of "If we base it on HTTP, we can use proxys and never have to worry about media" Of course this is not true and the amount of complexity that is put into each SIP device is much too great which is probably why Cisco prefers its own lighter "skinny" protocol. Sadly they own Sipura and Linksys and have SIP on their devices using countless hacks that make interop a nightmare. I am sure you can do many of these same attacks on any brand of phone. There are much better reasons out there to curse Cisco for being involved in VoIP. =D
 

++1 Yup, FreeSWITCH is great. Asterisk is powerful but leaks memory at high loads and in my real world experience doesn't handle 100+ calls well on a pure SIP setup on average hardware. I really enjoy the completely different way FreeSWITCH handles things. It feels more UNIX-y and hardened out of the box.

The biggest problem out there causing SIP toll fraud is people's extension passwords being set to things like "1234" in trixbox/asterisk/freepbx etc. It is "user error" but also "user ignorance" because these frontends and pbx software packages do not really bother telling their users to use secure extension passwords. In a perfect world extension passwords would be autogenerated to be very strong, but in FreePBX for example the extension password is called a 'secret' but since it is not called a password, I have had users set up Asterisk boxes with 1234 as the 'secret' who don't realize that actually means they are opening up UDP 5060 for connections to users who supply 1234 as their password. Hello, 50000+ calls to China/UK international DIDs! Goodbye, bank account :(

I do consulting for a VoIP PSTN gateway company and we are seeing a large amount of bruce force SIP registration attacks against our IPs all the time. We have implemented some DDOS protection to stop the abuse of our precious CPU cycles but the problem continues. It is more user related than Asterisk related. trixbox/freepbx/and the other frontends out there need to do more to make security their focus, rather than "omg look at the flash operator panel" and "look at the shiny bar graph showing live calls". Unfortunately the users of these frontends are not looking for that, so the shiniest frontend wins....

My current setup includes FreeSWITCH running on FreeBSD with the very nice FusionPBX frontend, which is based on the FreeSWITCH pfSense Firewall module. With these tools, a properly secured apache, and properly configured IP ACLs in place, I am seeing zero toll fraud on the system.

FreeSWITCH now has CODEC2 support just checked in. Please check it out! http://www.freeswitch.org/ /b

FYI,

Some have inquired as to using OpenBTS with FreeSWITCH as well as Asterisk. Alberto Escudero (aka AEP) wrote this wiki page nearly a year ago:

http://wiki.freeswitch.org/wiki/OpenBTS

It's slightly dated but the information is accurate.

-MC

We work on an open source softswitch called FreeSWITCH http://www.freeswitch.org/
Blocked ports and content filtering can mess up Voice over IP traffic running on your broadband line which can be used as a free alternative to the "Digital Phone" services many providers offer. Some entire countries already do this type of thing like China for instance. There are ways around it using secure packets so the payload cannot be sniffed and other workarounds but it would be a huge pain if we had to do that inside the US.

We have an Open Source project called FreeSWITCH http://www.freeswitch.org/ that allows you to do this sort of thing with any computer running Windows MAC or most UNIX. It can be paired with traditional phones with a small analog adapter or a hardware telephony card from Sangoma http://www.sangoma.com./ But you could just get a software phone for free as well and play around with it.
 

Having skype integrated into open source PBX [...] would be pretty good...

Asterisk supports Skype. As does FreeSWITCH.

hi, are you bad? do you know freeswitch? what is freeswitch? right, asterisk will to by atacked mor personaly i hate asterisk because there bad eventing engine, shell commands, CDR formats, SIP and other protocol stacks, Bad real time integration, very bad AMI and other unlimited bad stufs freeswitch mintin all that freeswitch run in win32, unix, linux, mac, *bsd, solaris, arm and other platform natively and will to by ported to all all all and all runing platforms thank to mikeJ the build and multi platform guru asterisk is bad because no win32 port anymore need cygwin the fact that #asterisk is full of hostile people hostility in #asterisk the fact that asterisk has deadlocks, and a hostile user community... "An example of the hostility in the asterisk community is how they treat Diego, and at's why Diego reacts against them, Diego is a very good person." freeswitch let you do ZRTP, SSL, TLS and TCP asterisk don't do TCP/TLS except for 1.6.X that is very nbad the freeswitch developer, anthony, is a very well very good person that helped me much mor about freeswitch including there flexible and realy nice dialplan, XML user directory, ldap integration and mor! because freeswitch is Smart and awesome, freeswitch mintin compatibility with asterisk by adding a asterisk dialplan module, but asterisk don't mintin anything about compatibility freeswitch let you do your dialplan in XML, Asterisk, YAML and other supported syntax for real time integration, i love the freeswitch integration including XMLRPC, XML_curl, the very good event sockett module and library and other integration asterisk do any users meeting? bad, bad, bad, bad and very bad freeswitch have a weekly organised big and nice meeting in any freeday asterisk community don't meet any asterisk users except for astricon that i hate it and i'm going to CluCon to meet my freeswitch friends and lovers so, about you, you are very bad because you don't read anything about freeswitch please go to http://www.freeswitch.org/ http://wiki.freeswitch.org/ and read mor and never talk about anyone because diego is a very nice and a very long time friend that give the freeswitch community mor help for free without any payment in #freeswitch IRC channel and all people love it, except for you that i don't love you because you hate him why the fs developer (anthony) don't talked about it? because diego is loving freeswitch and promoting it, unstid of the very bad disasterisk so pplease review your post befaure posting bad+bad=you

hi, are you bad? do you know freeswitch? what is freeswitch? right, asterisk will to by atacked mor personaly i hate asterisk because there bad eventing engine, shell commands, CDR formats, SIP and other protocol stacks, Bad real time integration, very bad AMI and other unlimited bad stufs freeswitch mintin all that freeswitch run in win32, unix, linux, mac, *bsd, solaris, arm and other platform natively and will to by ported to all all all and all runing platforms thank to mikeJ the build and multi platform guru asterisk is bad because no win32 port anymore need cygwin the fact that #asterisk is full of hostile people hostility in #asterisk the fact that asterisk has deadlocks, and a hostile user community... "An example of the hostility in the asterisk community is how they treat Diego, and at's why Diego reacts against them, Diego is a very good person." freeswitch let you do ZRTP, SSL, TLS and TCP asterisk don't do TCP/TLS except for 1.6.X that is very nbad the freeswitch developer, anthony, is a very well very good person that helped me much mor about freeswitch including there flexible and realy nice dialplan, XML user directory, ldap integration and mor! because freeswitch is Smart and awesome, freeswitch mintin compatibility with asterisk by adding a asterisk dialplan module, but asterisk don't mintin anything about compatibility freeswitch let you do your dialplan in XML, Asterisk, YAML and other supported syntax for real time integration, i love the freeswitch integration including XMLRPC, XML_curl, the very good event sockett module and library and other integration asterisk do any users meeting? bad, bad, bad, bad and very bad freeswitch have a weekly organised big and nice meeting in any freeday asterisk community don't meet any asterisk users except for astricon that i hate it and i'm going to CluCon to meet my freeswitch friends and lovers so, about you, you are very bad because you don't read anything about freeswitch please go to http://www.freeswitch.org/ http://wiki.freeswitch.org/ and read mor and never talk about anyone because diego is a very nice and a very long time friend that give the freeswitch community mor help for free without any payment in #freeswitch IRC channel and all people love it, except for you that i don't love you because you hate him why the fs developer (anthony) don't talked about it? because diego is loving freeswitch and promoting it, unstid of the very bad disasterisk so pplease review your post befaure posting bad+bad=you

Actually, I'd call Diego's personal experiences quite biased. His intro into the FreeSWITCH community was rather rough, but because FS did a better job in his scenario than Asterisk he scored points at his job and has been quite the cheerleader ever since.

For posterity's sake here's an interesting email thread from a suggestion Diego made, namely that YAML might be a better solution than XML for FS configs: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-June/004019.html

(BTW, when he suggested that we all thought he was nuts... in fact, we occasionally still think he's nuts but then again most of us in #freeswitch @ irc.freenode.net are a bit crazy.)

I can attest to Diego doing a good job of helping new ones get up and running, especially when they speak Portugese as a first language.

Just my $0.02 - do with it as you will.
-MC

Or freeswitch. Show me a dual-port version of said wart and I'll make it into a router.

When you create closed source code you have a much higher chance of flaws because your code can not tested nearly as much as open source can. As the leader of an open source project, FreeSWITCH http://www.freeswitch.org/ , I am fortunate to have a very large crowd of beta testers who help ensure our releases are as stable as they can be. If you are selling the application and never letting anyone see the source you run a very high risk of missing something in Q/A and releasing buggy software. When people pay for it the will get angry so I am not surprised such a suggestion is being made but I find it unpractical to enforce since if it "works right" is hard to judge in some cases besides maybe medical equipment or other situation where human lives are at stake. Blue screens of death are hardly an excuse to sue anyone.

Our project (FreeSWITCH) uses the MPL for the main application and BSD for satellite libraries that we create that can be used by other projects etc.

Once you decide to have open source code, it's more logical to stick with the fact that at least the core code is FREE and come up with ways to develop a product on top of it if you want to have something to sell. Otherwise it sounds like an "open source tax" and businesses do not like uncertainty. If they choose to use a code base they need to know it will always be available.

since it's open source, you can add 64-bit yourself. That's the whole point of open source.

The whole point of open source is for projects to work together and combine their efforts to make better software. As I said I am the author of an open source software. It has over 300,000 lines of code of it's own then a large list of dependency libs that added up account for about 2.5 million lines of code see: http://fisheye.freeswitch.org/browse/FreeSWITCH A library developer makes a library for other people to use. Adding 64 bit support to someone else's library is an exercise best left to the lead developer since it's his decision to support it or not.

I was looking at using v8 in our open source soft-switch/pbx/telephony application server FreeSWITCH http://www.freeswitch.org/

We currently are using spidermonkey from Mozilla and it has it's ups and downs in the scalability department since it was not designed for thousands of concurrent sessions in a single process. The documentation for v8 was impressive but sadly, 64 bit is not supported. It would be nice to get 64 bit supported so we could experiment further with it because it looks really well written.

We have pocketshinx working on windows, mac and linux in FreeSWITCH. http://www.freeswitch.org/ /b

Several years ago I wrote a javascript module for Asterisk open source PBX
More recently I added it for my own project FreeSWITCH ( http://www.freeswitch.org/ )

We actually also support LUA, Python, Perl, JAVA and MONO as ways to script telephony apps.

It's quickly becoming a great new way to prototype and deploy audio driven apps for your phone system.

Please read the comment more carefully. FreeSWITCH is not FreePBX nor Asterisk. It's Asterisk done right and then some. More info about FreeSWITCH is at http://www.freeswitch.org/ and joining #freeswitch at irc.freenode.net will get you in touch with a very helpful community ready to answer your questions. Try it, be amazed, contribute, and enjoy!

This could be very useful in projects like FreeSWITCH which is an Open Source project for building telephony applications. More info at http://www.freeswitch.org/

It's bad enough that a 2 million line program written from scratch would suddenly be infected by the GPL by including a 1 line file, but now we are supposed to propagate the virus with our web servers? I am perfectly happy to allow the GPL and its descendants to exist but I am reluctant see its roots dig deeper into the minds of ill-informed developers who do not realize the only goal of this license is to see how far it can spread. Before anyone tries to flame me be aware I am simply expressing my opinion, if you disagree, disagree with objectivity. I myself am an open source developer and I practice what I preach http://www.freeswitch.org/ I release my code under the MPL and BSD licenses which are actually much more liberal licenses than GPL but all the propaganda would have you believe otherwise. If you are going to write free code and give it away, then stop worrying about how other people are going to use it. When someone takes your code and makes a service out of it, don't you think they put any of their own work into it building an infrastructure etc?

Browsing the Truphone website I found their VoIP platform is about 90% open source: OpenSER, Asterisk, FreeSWITCH ... (see here). That's a great news for open source community, perhaps bad for telco vendors ...

http://www.freeswitch.org/ it can do most if not all the features they say are advanced in the conference.

Don't hold your breath.

I used to do quite a bit of stuff with Asterisk and I happen to run 7 production machines across a DS3 circuit each responsible for up to 4 T1 worth of traffic (92 calls in our current configuration of PRI with 1 DCHAN per span) I am somewhat skeptical about what the boxes would do if they really ran all 92 channels at once It think we have never had more than 2 of the T1's full on a given machine so it's a good thing we load balance them.

I decided I was not happy with this situation so I began work early this year on my own open source soft switch called
FreeSWITCH http://www.freeswitch.org/

As pointed out by someone else there are not very many details to go on in this article but I would venture to say the author's use of the term "Software vendors" implies he is talking about commercial distribution of software. That would suggest he wants companies who sell or license software to be responsible for it not necessarily the authors of the code.

If so, OSS contributors would not be risking anything unless they were also somehow licensing or selling the code for money. I run an open source project at http://www.freeswitch.org./ If someone turned my free code into a commercial product and started selling it, I would certianly want to see disgruntled customers suing *them* and not me =D

FreeSWITCH has srtp now. Check out www.freeswitch.org /b

The potential problem is that encryption of the voice stream adds latency to the transmission of the stream. Optimally you want 150 ms or less to pass in transmission, otherwise Bad Things can occur.

That being said, we have just switched Freeswitch to use SRTP in the past few days, which appears to support keyed transport. Does anybody else have experience using this library and can tell about your experience encrypting SIP and/or RTP with it?

from: http://www.freeswitch.org/docs/

"Licensing
Freeswitch is licensed under the terms of the MPL 1.1"

this license is *not* compatible with the gpl. even mozilla.org has stopped using this license:

Mozilla Relicensing FAQ
http://www.mozilla.org/MPL/relicensing-faq.html

mozilla is relicensing all of their code under a triple mpl/lgpl/gpl license in order to make their products compatible with the gpl. please consider doing the same with freeswitch.

read this if you need some more convincing as to why to relicense:

Make Your Open Source Software GPL-Compatible. Or Else.
http://www.dwheeler.com/essays/gpl-compatible.html

bottom line, if freeswitch isn't gpl-compatible it's much less likely to be successful.

My name is Anthony Minessale, After considerable contribution to Asterisk I have learned a great deal about telephony here is a list of my personal contributions to Asterisk: http://www.cluecon.com/anthm.html

The biggest lesson I have learned is that the fundamentals of Asterisk are built on assumptions and hard coded limitations. The flow chart for its code will make you dizzy:

http://www.freeswitch.org/astdoc/structast__channe l__coll__graph.jpg
http://www.freeswitch.org/astdoc/pbx_8c__incl.jpg

People who use asterisk from the outside wouldn't know there is absolutely no structure or discipline in the code and may not care. But once they invest a ton of time trying to make their dream Telco or whatever their dreams may be, the truth is all too obvious. Spoken from experience, only a seasoned technical wizard with years of computer skills to boast will ever be able to successfully implement Asterisk beyond a modest implementation. To truly understand how Asterisk works holds only a slightly smaller prerequisite. To those who find this unimportant, I understand your point, but be aware that Asterisk, being an open source project, needs to have a somewhat easy learning curve to attract new developers especially considering the developer turnover they suffer due to the maddening politics their community has to offer. The development is focused on owning all the code even if it means re-inventing things that already exist just to maintain the right to sell the code. This practice is fine with me though I am less than pleased by the end result when the home-rolled version is a poor contender with several existing solutions. The modular intentions of Asterisk are great though there is no structure there either. Any module can dig its way into nearly all of the code of the core and often, inexperienced module programmers will re-implement existing functionality to the extent that even inside the same C source file, you may find multiple versions of the same functions with different names. The other problem with Asterisk modules are that many of the in-tree modules carry cross dependencies that make it impossible for the core to function without them. Some modules even depend on each other. This practice limits the portability since many operating systems will not tolerate one dynamic object from using symbols from another without hard linking them together. This is not the worst offense as far as portability; there are dozens more with many being accredited to Linux-specific assumptions. Apart from the technology problems the biggest remaining problem to consider is the community. The first experience for most Asterisk newcomers is an IRC channel where people fight for supremacy like information hungry pirates hording what they know and then sticking it to people for being so "stupid". (In other words, in the same boat they were in a few months back.) For those of us who are experienced developers, we are used to the l33t thing. The deal breaker is the issue management process. Submissions will generally be ignored for months then a one sentence overview will command the developer to fix minor issues and resubmit. This is almost tolerable if the submitted code was a new feature but more times than not it also happens with meaningful clean-up and repair of broken core functionality. I have heard this same complaint from countless ex-asterisk contributors over the past year and I am sure it is the number one cause of their ex status.

In conclusion, I actively develop Asterisk code but now I only do it as a consultant. I am quite good at it and I know what I am talking about and I feel that the issues with Asterisk will never be addressed because there may be more Asterisk users every day but there are also less developers every day too and soon all the developers will be

My name is Anthony Minessale, After considerable contribution to Asterisk I have learned a great deal about telephony here is a list of my personal contributions to Asterisk: http://www.cluecon.com/anthm.html

The biggest lesson I have learned is that the fundamentals of Asterisk are built on assumptions and hard coded limitations. The flow chart for its code will make you dizzy:

http://www.freeswitch.org/astdoc/structast__channe l__coll__graph.jpg
http://www.freeswitch.org/astdoc/pbx_8c__incl.jpg

People who use asterisk from the outside wouldn't know there is absolutely no structure or discipline in the code and may not care. But once they invest a ton of time trying to make their dream Telco or whatever their dreams may be, the truth is all too obvious. Spoken from experience, only a seasoned technical wizard with years of computer skills to boast will ever be able to successfully implement Asterisk beyond a modest implementation. To truly understand how Asterisk works holds only a slightly smaller prerequisite. To those who find this unimportant, I understand your point, but be aware that Asterisk, being an open source project, needs to have a somewhat easy learning curve to attract new developers especially considering the developer turnover they suffer due to the maddening politics their community has to offer. The development is focused on owning all the code even if it means re-inventing things that already exist just to maintain the right to sell the code. This practice is fine with me though I am less than pleased by the end result when the home-rolled version is a poor contender with several existing solutions. The modular intentions of Asterisk are great though there is no structure there either. Any module can dig its way into nearly all of the code of the core and often, inexperienced module programmers will re-implement existing functionality to the extent that even inside the same C source file, you may find multiple versions of the same functions with different names. The other problem with Asterisk modules are that many of the in-tree modules carry cross dependencies that make it impossible for the core to function without them. Some modules even depend on each other. This practice limits the portability since many operating systems will not tolerate one dynamic object from using symbols from another without hard linking them together. This is not the worst offense as far as portability; there are dozens more with many being accredited to Linux-specific assumptions. Apart from the technology problems the biggest remaining problem to consider is the community. The first experience for most Asterisk newcomers is an IRC channel where people fight for supremacy like information hungry pirates hording what they know and then sticking it to people for being so "stupid". (In other words, in the same boat they were in a few months back.) For those of us who are experienced developers, we are used to the l33t thing. The deal breaker is the issue management process. Submissions will generally be ignored for months then a one sentence overview will command the developer to fix minor issues and resubmit. This is almost tolerable if the submitted code was a new feature but more times than not it also happens with meaningful clean-up and repair of broken core functionality. I have heard this same complaint from countless ex-asterisk contributors over the past year and I am sure it is the number one cause of their ex status.

In conclusion, I actively develop Asterisk code but now I only do it as a consultant. I am quite good at it and I know what I am talking about and I feel that the issues with Asterisk will never be addressed because there may be more Asterisk users every day but there are also less developers every day too and soon all the developers will be