Unfortunatlly everything goes quite well with Cisco if the entire network is only Cisco. We've managed by mistake to convince in several situations Cisco phones (7940) to reboot after a well formatted SIP packet(RFC 3261 compliant) that wasn't in the way Cisco thinks SIP should be. I think that free/open software is starting to be backed up by companies that are able to provide the technical support for any kind of issues. A few companies which do that are: Null Team which supports Yate, Digium which supports Asterisk.
Actually these days Yate supports conference, transfer (3 way calling if i remeber correctly is some sort of american conference - since Yate supports conference it also supports this). I'm sorry that the website is a little bit outdated and thank you for letting us know about that problem.
Actually most of the women i know care about what a guy knows to do and how his body looks like. Women don't care much what men are wearing. What most of the IT guys miss is some gym classes not fancy clothes. Clothes are important just for other men especially for older since this is their only way to show that "they still have the move".
I think the corect phase was " comparing with Nokia SIP stack, YASS is a piece of art". I didn't mention OPAL since is trying to use SIP as H.323 and i didn't mention reSIProcate (VOCAL and sipX come from the same people) and the other since i don't have a reason to do mention them. Is very disapointing to see that insted of being a winning for free software nokia open source site is just a way for nokia to gain for free some developers and beta testers.
I've just took a look at then Sofia-SIP stack. One of the most horribile pieces or code i've saw lately. I mean even oSIP which is the most rubbish SIP stack from the free world look way better than this. I won't compare it with YASS (Yate SIP stack) which is a piece of art if you compare it with SIP stack. I can't belive that in this days someone will write code in the way Sofia-SIP is writte. Just compare how complicated it is.
Maybe i'm wrong but Linux has a monolitic kernel and Windows is the one who have a micro kernel. So from my point of view, sipX is pretty much like Windows with a microkernel and Asterisk is like Linux with a monolitic kernel. Just a point from a Yate (http://yate.null.ro)developer. We have our own SIP stack which was written in 3 weeks, maybe is not full but is working, and is usable in embeded enviroments like uclibc. You see, the point is not if IAX is better or worse then SIP (personaly i like H.323 - mainly because Craig from OpenH323 is a cool guy), but the point is to support all protocols, to not lock the user. Asterisk dosen't support H.323 but at least (good or worse) it gives also IAX support (don't foeget about MGCP and others) which solve some problem. sipX is an elitist system which dosen't try to solve something but rather tries to force people to use SIP, which is agains free software (i'm also a Richard Stallman fan) way of doing things. Yate took this to the next level trying also to do a modular system that support all the protocols. If you don't have a managable PBX you may not use it in most of the cases, and asterisk can't be managed (that mean good cdr, routing, remote control, modular system). I don't know about sipX since i wasn't able not even to compile it.
Diana
P.S. I appologize for my english mistakes. I hope reader will pay more attention to ideas insted of language.
Asterisk is not the only software who can do IAX-SIP. Yate can do it also. Yate has nothing to do with Digium hardware. It support "and" Digium hardware but that's all. And IAX products are mainly at the same price as SIP. I'm not a fan nither for IAX, nither for SIP (i like H.323:)), but you can deal with both of them quite decent.
It seems that i've end up with a stack that have 90 K of code and is flexibile enough to be used for a client, or a server or a proxy in a single or multi threaded program. Is called YASS (Yet Another SIP Stack), and when is compiled it actualy have something like 83k now (of course we will develop it). I've try sipfoundary in the past and i know how huge is it. And the test i've told you about have been for a SIP - H.323 signalling proxy not for pure SIP. For pure SIP, SER which is a very good SIP router can handle 40 calls on an embedded systems. I still didn't see your answer regarding the size of the sipX. Actualy one year ago i have download the CVS and i have try to compile sipX. After getting 2 GB of my free space the compilation crushed. I have never seen such a horribile way of abusing C++ like that project - exceptions, template - many levels, you name it. Is all there.
I'm a developer for Yate (http://yate.null.ro) and i have try to compile that sip stack. After getting to 2 Gb i have quit. I have look for like 6 months for a good SIP stack, we've ended by doing our own SIP stack. You want to know why? Because it dosen't crush. In fact one of Yate jobs was to work as a H.323-SIP signalling proxy, because sipfoundry stack didn't manage to route more then 15 calls. What are we talking about here? About telephony which must be stable or about some Windows game?
Actualy a gateway will be the ATA in that case and not sipX. sipX will be just a sip router and that's all. In fact i doubt that sipX can be ever called a PBX or a gateway.
You got me wrong. The problem is not the hardware. The question was : "Can you handle more the 30 calls stable with Asterisk?". And yes maybe is trolling, but before going to buy something you should see both reviews not only the good ones.
Yate (Yet Another Telephony Engine) is also a gateway and a PBX. It supports H323 (much better then asterisk), SIP (with a nice stack that it can be actualy reused), IAX2 (with a forked version of libiax2), and ISDN (PRI and BRI) using zaptel drivers. The best part is that is much more flexibile then any other similar project around. Is not like sipX just SIP based, and is not like Asterisk a emulation of PSTN over VoIP. Is a real VoIP server that actualy deal also with PSTN.
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Unfortunatlly everything goes quite well with Cisco if the entire network is only Cisco. We've managed by mistake to convince in several situations Cisco phones (7940) to reboot after a well formatted SIP packet(RFC 3261 compliant) that wasn't in the way Cisco thinks SIP should be.
I think that free/open software is starting to be backed up by companies that are able to provide the technical support for any kind of issues. A few companies which do that are: Null Team which supports Yate, Digium which supports Asterisk.
Actually these days Yate supports conference, transfer (3 way calling if i remeber correctly is some sort of american conference - since Yate supports conference it also supports this). I'm sorry that the website is a little bit outdated and thank you for letting us know about that problem.
Actually most of the women i know care about what a guy knows to do and how his body looks like. Women don't care much what men are wearing. What most of the IT guys miss is some gym classes not fancy clothes.
Clothes are important just for other men especially for older since this is their only way to show that "they still have the move".
I think the corect phase was " comparing with Nokia SIP stack, YASS is a piece of art".
I didn't mention OPAL since is trying to use SIP as H.323 and i didn't mention reSIProcate (VOCAL and sipX come from the same people) and the other since i don't have a reason to do mention them.
Is very disapointing to see that insted of being a winning for free software nokia open source site is just a way for nokia to gain for free some developers and beta testers.
I've just took a look at then Sofia-SIP stack. One of the most horribile pieces or code i've saw lately. I mean even oSIP which is the most rubbish SIP stack from the free world look way better than this.
i b/ysip/ - Yate SIP stack
/ - oSIP
o fia-sip/ - Sofia-SIP
I won't compare it with YASS (Yate SIP stack) which is a piece of art if you compare it with SIP stack.
I can't belive that in this days someone will write code in the way Sofia-SIP is writte. Just compare how complicated it is.
http://voip.null.ro/cgi-bin/cvsweb.cgi/yate/contr
http://savannah.gnu.org/cgi-bin/viewcvs/osip/osip
http://cvs.sourceforge.net/viewcvs.py/sofia-sip/s
I think in the end that what Nokia did was just to throw some rubbish code arround hoping to get some more bug fixes.
You can actualy use Yate as a SIP - IAX proxy and it also work on embedded systems like Linksys Wireless Access Point, or any other uclibc device.
Maybe i'm wrong but Linux has a monolitic kernel and Windows is the one who have a micro kernel. So from my point of view, sipX is pretty much like Windows with a microkernel and Asterisk is like Linux with a monolitic kernel.
Just a point from a Yate (http://yate.null.ro)developer. We have our own SIP stack which was written in 3 weeks, maybe is not full but is working, and is usable in embeded enviroments like uclibc.
You see, the point is not if IAX is better or worse then SIP (personaly i like H.323 - mainly because Craig from OpenH323 is a cool guy), but the point is to support all protocols, to not lock the user. Asterisk dosen't support H.323 but at least (good or worse) it gives also IAX support (don't foeget about MGCP and others) which solve some problem. sipX is an elitist system which dosen't try to solve something but rather tries to force people to use SIP, which is agains free software (i'm also a Richard Stallman fan) way of doing things.
Yate took this to the next level trying also to do a modular system that support all the protocols. If you don't have a managable PBX you may not use it in most of the cases, and asterisk can't be managed (that mean good cdr, routing, remote control, modular system). I don't know about sipX since i wasn't able not even to compile it.
Diana
P.S. I appologize for my english mistakes. I hope reader will pay more attention to ideas insted of language.
Asterisk is not the only software who can do IAX-SIP. Yate can do it also. Yate has nothing to do with Digium hardware. It support "and" Digium hardware but that's all. And IAX products are mainly at the same price as SIP. I'm not a fan nither for IAX, nither for SIP (i like H.323 :)), but you can deal with both of them quite decent.
Asterisk is pretty much like Windows in fact. Is big, bloated and unstable. sipX is like Asterisk father - big+, bloated+,unstable+.
It seems that i've end up with a stack that have 90 K of code and is flexibile enough to be used for a client, or a server or a proxy in a single or multi threaded program. Is called YASS (Yet Another SIP Stack), and when is compiled it actualy have something like 83k now (of course we will develop it). I've try sipfoundary in the past and i know how huge is it. And the test i've told you about have been for a SIP - H.323 signalling proxy not for pure SIP. For pure SIP, SER which is a very good SIP router can handle 40 calls on an embedded systems.
I still didn't see your answer regarding the size of the sipX. Actualy one year ago i have download the CVS and i have try to compile sipX. After getting 2 GB of my free space the compilation crushed. I have never seen such a horribile way of abusing C++ like that project - exceptions, template - many levels, you name it. Is all there.
I'm a developer for Yate (http://yate.null.ro) and i have try to compile that sip stack. After getting to 2 Gb i have quit. I have look for like 6 months for a good SIP stack, we've ended by doing our own SIP stack. You want to know why? Because it dosen't crush. In fact one of Yate jobs was to work as a H.323-SIP signalling proxy, because sipfoundry stack didn't manage to route more then 15 calls. What are we talking about here? About telephony which must be stable or about some Windows game?
Actualy a gateway will be the ATA in that case and not sipX. sipX will be just a sip router and that's all. In fact i doubt that sipX can be ever called a PBX or a gateway.
You got me wrong. The problem is not the hardware. The question was : "Can you handle more the 30 calls stable with Asterisk?".
And yes maybe is trolling, but before going to buy something you should see both reviews not only the good ones.
I wonder how often do you have to restart your Asterisk? And i also wonder if you can deal with more then 30 calls?
Yate (Yet Another Telephony Engine) is also a gateway and a PBX.
It supports H323 (much better then asterisk), SIP (with a nice stack that it can be actualy reused), IAX2 (with a forked version of libiax2), and ISDN (PRI and BRI) using zaptel drivers.
The best part is that is much more flexibile then any other similar project around. Is not like sipX just SIP based, and is not like Asterisk a emulation of PSTN over VoIP. Is a real VoIP server that actualy deal also with PSTN.