sorry, missed the question. the sipXTAPI, which includes the stack, media processing, call processing and the API is about 850k and the stack alone is about 500k.
Anyway, you insist on being insulting w/out out being productive. Best of luck
thanks for clarifying. the stack actually runs at about 40 calls per second now and is completely stable. the reSIPprocate stack is also on SIPfoundry, is used for high-performance session border controllers, among other applications, and is also fast and stable.
Of course, these stacks may not have been fast or light enough for your particular needs, and so you have built your own. Great. But that does not reflect on the quality and value of stacks for other applications, nor of the sip PBX, proxy, softphone or UA applications, so the "Windows game" comment is maybe a little bit gratuitous.
It might also have been useful to you work w/in the SIPfoundry community to build the stacks you needed. You may have ended up w/ what you wanted with broad community support?
Perhaps there avenues for collaboration? Shall we go off-line and discuss? ossip99@yahoo.com
yes, documentation, like Rome, is not built in a day. However, if you are after something specific pls. post of the sipX dev list. You'll get info right away.
You are certainly corect that the ATA is a gateway, just like a Digium card is a gateway, and that sipX is not a gateway, it is a Linux-based, SIP, PBX and proxy. That, by the way, is the value of the SIP architecture. It works w/ any SIP end-point - gateway, phone, etc. And in SIP, the end-points (phones, gateways, applications)are intellegent. Some featurs live in the phones, and phone can broadcast their state (aka presence) to other applications. That means every PBX feature does not need to be implemented in a central server. That and the plug & play nature of SIP is the reason every serious telephony company is implementing SIP as the core of the network. sipX works w/ Cisco, Vegastream, Mediatrix, AudioCodes gateways, and has been tested w/ Digium, Quintum and and Citel gateways. More to come. Phones suppoted include Cisco, Polycom, SIPura, with testing underway for Grandstream, Xten, SIPura ATAs, Snom, and Astra phone.
Some features are not implemenetd in end-points though, but rather as server components, like voice mail. Those are are in SIP servers.
re: the comment that sipX is not a PBX is simply wrong. It certainly is not built in the old TDM command and control model - they way TDM PBXs are built and VoIP systems from Avaya, Cisco et al are built. It is absolutely a PBX that implements the SIP architecture and provides all the features of a PBX. Happens to be a proxy too, which provided inter-system rounting for VoIP and applications, etc. You really should have a closer look at www.sipfoundry.org. It is just possible that the state of the art moved forward a bit
You might consider looking at little more closely at www.sipfoundry.org. What you will find is that sipX is configuarble in the way you describe, using xml. You can tweak all the nobs. However, some classes of users like a management tool - not bad just different, recognizes that there are a lot of players who might open source attractive
The Windows vs. Linux analogy is off-base.
Actually not so. sipX is a fully functional PBX that supports SIP phones, and analog phones through an ATA. Support is also in development for supporting the Citel gateway, which means sipX will support Nortel digital PBX phones.
the sipX family includes a PBX (SIP proxies, SIP media server, and browser-based configuration system); SIP routing proxy (same pieces, less the media server), SIP spftphone and a SIP UA
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sorry, missed the question. the sipXTAPI, which includes the stack, media processing, call processing and the API is about 850k and the stack alone is about 500k. Anyway, you insist on being insulting w/out out being productive. Best of luck
thanks for clarifying. the stack actually runs at about 40 calls per second now and is completely stable. the reSIPprocate stack is also on SIPfoundry, is used for high-performance session border controllers, among other applications, and is also fast and stable. Of course, these stacks may not have been fast or light enough for your particular needs, and so you have built your own. Great. But that does not reflect on the quality and value of stacks for other applications, nor of the sip PBX, proxy, softphone or UA applications, so the "Windows game" comment is maybe a little bit gratuitous. It might also have been useful to you work w/in the SIPfoundry community to build the stacks you needed. You may have ended up w/ what you wanted with broad community support? Perhaps there avenues for collaboration? Shall we go off-line and discuss? ossip99@yahoo.com
yes, documentation, like Rome, is not built in a day. However, if you are after something specific pls. post of the sipX dev list. You'll get info right away.
You are certainly corect that the ATA is a gateway, just like a Digium card is a gateway, and that sipX is not a gateway, it is a Linux-based, SIP, PBX and proxy. That, by the way, is the value of the SIP architecture. It works w/ any SIP end-point - gateway, phone, etc. And in SIP, the end-points (phones, gateways, applications)are intellegent. Some featurs live in the phones, and phone can broadcast their state (aka presence) to other applications. That means every PBX feature does not need to be implemented in a central server. That and the plug & play nature of SIP is the reason every serious telephony company is implementing SIP as the core of the network. sipX works w/ Cisco, Vegastream, Mediatrix, AudioCodes gateways, and has been tested w/ Digium, Quintum and and Citel gateways. More to come. Phones suppoted include Cisco, Polycom, SIPura, with testing underway for Grandstream, Xten, SIPura ATAs, Snom, and Astra phone. Some features are not implemenetd in end-points though, but rather as server components, like voice mail. Those are are in SIP servers. re: the comment that sipX is not a PBX is simply wrong. It certainly is not built in the old TDM command and control model - they way TDM PBXs are built and VoIP systems from Avaya, Cisco et al are built. It is absolutely a PBX that implements the SIP architecture and provides all the features of a PBX. Happens to be a proxy too, which provided inter-system rounting for VoIP and applications, etc. You really should have a closer look at www.sipfoundry.org. It is just possible that the state of the art moved forward a bit
there is a Gentoo build at http://wiki.calivia.com/index.php/Main_Page
so - great idea. any interest in working on that? I think it would be great to get this going. Mind posting it on the sipx dev mailing list?
You might consider looking at little more closely at www.sipfoundry.org. What you will find is that sipX is configuarble in the way you describe, using xml. You can tweak all the nobs. However, some classes of users like a management tool - not bad just different, recognizes that there are a lot of players who might open source attractive The Windows vs. Linux analogy is off-base.
Actually not so. sipX is a fully functional PBX that supports SIP phones, and analog phones through an ATA. Support is also in development for supporting the Citel gateway, which means sipX will support Nortel digital PBX phones. the sipX family includes a PBX (SIP proxies, SIP media server, and browser-based configuration system); SIP routing proxy (same pieces, less the media server), SIP spftphone and a SIP UA