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Written by Administrator Saturday, 03 May 2008 For almost a year now Nortel engineers contributed significant components and code to the SIPfoundry sipXecs project. The SIPfoundry sipXecs project was selected by Nortel because of its solid and scalable architecture. The idea was to assess whether Nortel's next generation unified communications solution should be based on the sipXecs open source project. Nortel now announced their new unified communications solution called Software Communications Server SCS 500 on April 10th. The SCS500 product is completely based on the sipXecs code base. In addition to the Nortel channel the SCS500 product is sold by Dell as well as IBM.
This is big news for us of course. It is not that often that a tier 1 player like Nortel adopts a solution that is developed by a third party. And it is not that often that a big player like Nortel adopts an open source project. Nortel is fully committed to contributing to the open source effort going forward. There are many exciting things coming in the next release. Among them conferencing, full SIP trunking and NAT traversal. Tag it:       |
Written by Administrator Sunday, 16 March 2008 Over 170 new features and improvements later, sipXecs Release 3.10 is now stable and released. It can be downloaded from the Downloads page. Release 3.10 is our biggest release to date adding more new features and improvements and addressing more issues than ever before. The developer team has pushed hard to make sipXecs even easier to install and use than it already was before. Plug & play management got enhanced with many new devices supported. With release 3.10 sipXecs also improved on scalability, robustness and interoperability stretching the upper limit in terms of supported users to 10.000. A list of new features and capabilities is available on the Wiki In addition the roadmap on the Wiki provides an outlook into the next release. Development for sipXecs 4.0 has already started. Major new features include:
Thanks to everyone who contributed to this release! Tag it: |
Written by Administrator Saturday, 09 February 2008 Only a few monts after the stable release of 3.8, release 3.10 is almost ready.
So far we added 162 new features and addressed over 300 bugs and other issues in the upcoming release 3.10 (up from 115 new features and 238 bugs and issues addressed for release 3.8). An overview in pdf format can be downloaded following this link .
Release 3.10 is our biggest release so far with the most new features since the sipXecs project was started. Our roadmap gives you an idea of what's new and there is also a detailed release notes document. We are expecting to announce a release candidate 1 (RC1) of the 3.10 release by early March. The screenshot above shows the new phone and gateway auto-discovery function. It scans your network and auto-discovers all the devices, which makes installation even easier than it already was. Pre-release software can be downloaded. The release has now been in extensive QA for three months and is very stable. Tag it: |
Written by Administrator Saturday, 05 January 2008 sipXecs Release 3.8.1 is now stable and released. It can be downloaded from the Downloads page. Release 3.8.1 is the first maintenance release for the stable release 3.8. Because we did not post an announcement when 3.8 came out, the following summarizes some of the most important new features.
The major focus for release 3.8 was to add features and continue to improve ease-of-use, robustness and quality of the solution. Major new features include:
Thanks to everyone who contributed to this release! Tag it: |
Written by Administrator Saturday, 14 April 2007 Pingtel launches new sipXecs project New ACD Call Center Server and new Presence Server now available in open source as part of the sipXecs project What does this mean for me? See additional details in the post made to the developers and users mailing list by the project maintainers.
Effective today, the maintainers of the sipXpbx project and Pingtel
Corp. have launched sipXecs, a branch of the sipXpbx project, which
represents a major upgrade to the sipX family and includes features and
functionality not previously available in sipXpbx. 'ecs' stands for
"Enterprise Communication System", which reflects the fact that sipX is
now much more than just a phone system (pronounce the new name
'sip-eks-ee-see-es' or 'sip-eks-eks'). Tag it: |
Written by Administrator Thursday, 12 April 2007 sipXecs wins Miercom / NetworkWorld SMB open source IP PBX test contest against three Asterisk based solutions. To quote NetworkWorld: 
"Pingtel's SIPxchange earns the Clear Choice award for triumphing over the field in our endpoint interoperability and architecture categories. In the latter category we examined how the product was designed to work. SIPxchange comprises some of the more common practices found in larger, proprietary systems, such as direct paths for the media streams. This limits the burden on the server and allows for better scalability and reliability." See the test results here ! Tag it: |
Written by Administrator Thursday, 15 February 2007
sipx Release 3.6 is now stable and released. It can be downloaded from the Downloads page. This release represents a major milestone for the project. With release 3.6 the sipX solution has become much more robust thanks to a lot of performance and stress testing done over the last three months. Real deployments serve in excess of 5,000 users using a single redundant server and we are now comfortable that the sipX solution can scale up to about 10,000 users. The High-Availability implementation has been hardened and its administration was further automated. Also, the sipX Configuration Server has made lots of progress further improving on plug & play management of phones and overall ease of use. Major new features includde:
Thanks to everyone who contributed to this release! Tag it: |
Written by Administrator Saturday, 06 January 2007 
Bring these buttons to life! sipX 3.7, the current main line, now supports softkeys for speed dial and directory. Using the user portal of sipXconfig every user can define personal speed dial keys. These settings are projected to every phone the user is using. Directory information is generated by sipXconfig based on the users created in the system. In addition, spreadsheets with additional external users can be uploaded. You can create as many different directories you like and assign them to users or groups of users. We have built a generic model that is applicable to most if not all phones supporting sof- or hard keys for speed dial and directories. Currently only the phone model for Polycom phones supports this new capabilities. However, adding this generic capabilities to other phone models is pretty easy. If you'd like to know more about it, ask on the list and you will get some help. As a next step we plan on adding Busy Lamp Field (BLF) capabilities to the speed dial screen. This will allow the status of other lines to be monitored on the phone. It is based on a new Resource List Server (similar to a presence server) that phones can subscribe to in order to get status of other lines. More info is on the Wiki .
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Written by Administrator Saturday, 06 January 2007
Installing sipXecs can be done in two different ways: a) using a sipXecs installation CD, which installs everything including the operating system, or b) using application RPMs that can be installed after the operating system is already running using packet management. The installation CD is based on CentOS and features automated installation of both the application and the operating system. System configuration is done using a graphical configuration wizard. The installation wizard allows automated configuration of a high-availability system with two servers operating in load-sharing redundant mode. In addition, the installation wizard offers to configure additional network services such as DHCP, DNS, and NTP on the host. sipXecs requires correct configuration of these services in order to operate properly. If external DHCP, DNS and NTP services are used make sure you follow the directions on the Wiki on how to set them up. As a new feature in the 3.10 release sipXecs offers a built-in test suite invoked from the Web administration UI that allows testing of properly configured network services. Use it to make sure everything is OK. Minimum system requirements for a production server are P4 or Core 2 Duo CPU, 1 GB RAM, 80 GB HD. Test systems can work with less. The server needs to be given a fixed IP address and hostname that resolves in DNS. Usually there are no special requirements imposed on the LAN infrastructure. Setting up QoS or VLANs is usually not required, but you do need a 100 Mbit/s switched environment. A wired infrastructure is clearly superior to wireless (WiFi), unless you have a very new WiFi deployment based on the 802.11n standard. The 802.11g standard only provides 54 Mbit/s peak shared across all users connected to the access point and only half-duplex. The sipXecs installation CD installs a standard CentOS based system. System administration after installation can be done using all the usual procedures and tools to administer such a system. In particular, updates to the system's operating system and the sipXecs application are done with the standard package management system. The sipXecs application includes logic to provide for automated upgrades between major releases (i.e. database migration and other configuration). Tag it: |
Written by Administrator Monday, 06 November 2006 Creating a new Skin for sipXecs is easy - It is fully skinnable
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Written by Administrator Monday, 30 October 2006 Plug & Play supported IP Phones Making or receiving a call combining a SIP phone with an IP PBX usually works out of the box. Providing reliable support for all the calling features, softkeys, presence capabilities, firmware management, profile generation, and auto-discovery capabilities is hard. It is especially hard if you aspire to do regression testing against new firmware revisions in phones and gateways as vendors make changes, add features, resolve bugs and introduce new ones. Based on our experience gained over the last many years we learned that a close relationship with the respective vendor is required to make this work in a reliable way. In order to facilitate cooperation with the vendor community we created and operate the industry's leading SIP interoperability test portal at http://interop.pingtel.com . Open to everyone, the portal is used to run automated tests for most critical features, which gives you an immediate view on how good a phone really is. Once you pass all the tests, creating a plug & play management plugin is easy and will yield very satisfying results. The following table provides some insight into what phones are of high quality and supported well by sipXecs. To a large extent this is a matter of testing and since the core developer team of sipXecs can only test that many phones on a regular basis, many other phones are community maintained.
We currently reccommend to stay with Polycom, Snom, or LG-Nortel and Audiocodes if you are looking for an easy deployment. Softphones:
Video phones and other codec related questions: With sipXecs media does not go through the sipXecs server but is routed direct on your LAN either between phones for internal calls or between a phone and a gateway for external calls. Therefore, sipXecs supports any codec the two participating end-points can negotiate. There is no limit on video calls other than the available bandwidth on your LAN. Polycom HD-Voice is supported as is every other high or low bitrate codec, provided the participating end points support it. sipXecs supports plug & play management for Nortel 1535 video phones as well as the Grandstream GXV-3000. In addition, sipXecs supports Counterpath softphones with video. How to add support for plug & play management for a new phone or gateway? A lot was written already about the ability to manage phones and gateways plug & play. The Wiki includes detailed documentation on what phone and gateway models are already supported and how you can add support for new ones. Again, we are using a plugin framework so that little to no programming is required to add support for a new phone. Essentially what is required is writing an XML specification file that describes all the possible parameters the phone can accommodate. The sipXconfig user interface is then auto-generated from that specification file.
Note: Every SIP compliant phone or gateway works with sipXecs. If it is not plug & play managed then it has to be manually configured, which is what you will have to do with most other IP PBX systems anyway. Tag it: |
Written by Administrator Monday, 30 October 2006 In the current development release (3.7) Dale added support for redirector plugins. This greatly enhances flexibility of the call routing framework as it is used by the registrar. Redirector plugins are consulted in a configurable order for every cal. ENUM and ISN dialing are implemented as a redirector plugin as are all the other pre-configured dialing rules that exist in the system. New ones can easily be added by writing a new plugin. The advantage of the plugin framework is that you do not have to understand the registrar and all the related complexities. Writing a redirector plugin follows a defined API that is much easier to understand. Tag it: |
| User Interface Translation Written by Administrator Monday, 30 October 2006 |
| Localization of the dial plan Written by Administrator Sunday, 29 October 2006 |
| Localization of Voice Prompts Written by Administrator Sunday, 29 October 2006 |
Written by Administrator Monday, 30 October 2006 sipXecs - A native SIP based Unified Communications Solution
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