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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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SIPfoundry

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SIPfoundry


Web site: http://www.sipfoundry.org - Wiki: sipX Wiki

SIPfoundry is a not for profit open source community. Our mission is to promote and advance SIP-related Open Source projects. Through SIPfoundry the users, developers, and distributors of SIP based products can collectively support each other and accelerate the growth and adoption of SIP.

Founded in March 2004, SIPfoundry established close ties with the SIP Forum as well as the IETF. SIPfoundry actively promotes the standardization of SIP, interoperability of SIP products and solution across the industry through the SIP Forum Test Framework (SFTF), as well as reference SIP implementations of key technology such as the reSIProcate SIP stack.

SIPfoundry is also the place where the development of sipX, The SIP PBX for Linux, takes place. This project aims at commoditizing the PBX functionality by offering a fully featured, standards compliant, and easy to use SIP PBX for free as an open source solution. We would like SIP to become part of the Internet the same way HTTP, SMTP, and XML became ubiquitous and drove rapid adoption of new services across the Internet.

The SIPfoundry community stands behind the following key projects and technologies:

1. The sipX Project

The sipX project is about developing the most feature rich and standards compliant SIP communications infrastructure for Enterprise use in a community organized open source effort. The sipX architecture is modular and consists of three main building blocks:
  • sipX Communications Server
  • sipX Media Server
  • sipX Configuration Server
While sipX packages these components to function as a SIP PBX, each server can also be used standalone.

2. The sipXphone, sipXezPhone, and sipXtapi UA SDK project

SIPfoundry provides a very comprehensive set of technologies for the development of SIP user agents supporting SIP voice communications, presence and instant messaging. sipXphone is a high functionality SIP softphone available both for Windows and Linux platforms that is derived from the once famous Pingtel xpressa phone. Recent additions include STUN support and a new GUI interface. sipXtapi is an easy to use comprehensive software development kit (SDK) for the development of all kinds of stand-alone or integrated SIP client solutions. sipXezPhone is a simple implementation of a SIP softphone using the sipXtapi SDK.

3. The reSIProcate Project

The reciprocate project is developing an object oriented SIP stack written in C++ and intended to serve as the SIP reference implementation. Developed by many of the same people who participate in the IETF�s standardization effort of SIP, the reSIProcate stack is fully standards compliant, feature complete, and often serves as a test environment for new initiatives such as the most recent IETF draft proposals for SIP security.
The reSIProcate stack is used in an increasing number of commercial products by companies such as TelTel, Jasomi, Attractel and Xten. Its design objective is to create a well documented and easy to use SIP stack for use in phones, gateways, SIP proxies, back-to-back user agents as well as instant messaging and presence applications.


4. The SIP Forum Test Framework (SFTF) Project

Interoperability of SIP products and SIP-based technologies is key to achieving SIPfoundry's objectives. The scope of SIPfoundry's initiatives to encourage and promote interoperability of SIP products and SIP-based technologies include the following:
  • Establishment of a test framework (SFTF),
  • Active participation in SIP based IETF standards development,
  • Involvement in forums (SIP Forum), and
  • Promotion of SIP testing interoperability events (SIPIT and SIMPLEt).

5. The Open Settlment Protocol (OSP) Client Toolkit Project

The OSP client Toolkit is a complete development kit for software developers who want to implement the client side of the European Telecommunication Standards Institute's Open Settlement Protocol (ETSI TS 101321). The OSP client Toolkit includes source code written in ANSI C, test tools and extensive documentation on how to implement OSP. A hosted OSP test server is freely available on the Internet for all developers to test their OSP implementation.

OSP is an Operational Support System (OSS) protocol well suited for managing inter-domain routing, access control and accounting of SIP transactions. OSP uses the communications protocols below to convey messages. The content of an OSP transaction is an HTTP message formatted according to the standard for MIME. Individual components in the message are XML documents and the message may be signed with an S/MIME digital signature.


5. The Message Session Relay Protocol (MSRP) Project

This project is still in an early phase of development and aims to create a reference implementation of an MSRP stack. We are currently targeting the most recent drafts of the base specification and the relay extensions. The current plan is to create a beta-quality stack suitable for use in a client, and then add relay functionality (for both clients and relays) after the basic functionality is in place. Work is done in close cooperation with the IETF and the MSRP implementation will serve as a first reference implementation.

Goals of this project include:

  • Validation of the implementability of the existing IETF specifications for MSRP.
  • Creation of a freely available stack against which others can test their implementation.
  • Creation of a stack that can be integrated into both opensource and proprietary products to add MSRP functionality.


Additional Resources


Created by oej, Last modification by joachim on Sat 20 of Jan, 2007 [11:48 UTC]

Comments Filter

by Craig Lawrence on Tuesday 13 of February, 2007 [11:03:20 UTC]

by Craig Lawrence on Tuesday 13 of February, 2007 [11:02:18 UTC]
We may be able to provide you with a solution if you are looking for a call centre interface with secure login for agents, power dialler, call data import and export..... SIP / global deployment / call recording on demand supported and FTP upload / dispositions and call back / this is a fully supported and hosted solution with seperate partition available

virtual call centre

by manoj on Tuesday 12 of December, 2006 [01:06:16 UTC]
WE are looking for Hosted Call centre or Virtual Call centre, through a SIP Account, globally, ........... any one can help ..........

hosted call center

by manoj on Tuesday 12 of December, 2006 [01:03:22 UTC]
we are looking for hoster call centre or virtual call centre throught SIP account golbally.... is it possible over Wide area Network.. 

SIP over TCP?

by Gia Nguyen on Monday 24 of April, 2006 [19:08:23 UTC]
Hi there,

We currently are using UDP for our SIP implementation but are looking at TCP due to firewall restrictions outside of our control (specifically some sort of RTP tunneling via TCP port 80). Anyone has experiences with this? Pros/Cons? Any feedback would be appreciated.

Thanks ahead of time. Cheers,

Gia Nguyen
gnguyen@telecontinuity.com


we are lokking for....

by reza on Saturday 04 of February, 2006 [12:08:03 UTC]
we are looking for a linux base programmer for inmoving our own sip proxy , please contact me at grkashani@yahoo.com

Re: Explain this

by bigian on Thursday 15 of December, 2005 [11:22:08 UTC]
Rigthto that would explain it

Section 2 the SipXtapi link goes no where

Re: Explain this

by admin on Wednesday 14 of December, 2005 [19:39:24 UTC]
Double clicking on a page is a short cut to get to edit mode.
I don't see any link for Download, where do you see that link?

Explain this

by bigian on Wednesday 14 of December, 2005 [13:30:33 UTC]
If I double click on the SipXTapi link I get to edit the front page???????

  • Edit* OK I see the link to let me edit the page however why does the link to download and API + several other links on this page take me to the editor if I double click them? Also why do they go nowhere when you just single click them???

Link

by bigian on Wednesday 14 of December, 2005 [11:30:59 UTC]
See this http://img228.imageshack.us/my.php?image=image19zu.jpg

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