OpenVox Communication Co. Ltd, a global leading provider of the most advanced open source Asterisk® telephony hardware and software products, has announced today to release the 3rd generation asterisk-based appliance — IX130 to the open source telephony community. The IX130 comes with OpenVox creative modular design that provides two plug-in modules which enable different telephony interfaces connection to the new IPPBX. Equipped with OpenVox IPC110 embedded motherboard, the IX130 provides cost-effective and stable solutions to SMBs with an attractive price.

The IX130 is a standard 19” 1U appliance. It allows users to integrate up to 2 pieces of OpenVox PCI-E telephony interface cards, with any combinations of analog, BRI, PRI, GSM or transcoding interfaces. The optional HDD case provides the possibility for hot plug function to the storage device. Thus users can simply replace a backup HDD whenever needed without opening the appliance. What’s more, the IX130 provides two flexible ways of power supply: internal and external and it’s up to you to select one of them to power up the IX130.

“We’ve been developing asterisk appliances for SMBs for years and we know the market well. With sufficient feedbacks from our customers and our marketing surveys, we are proud to announce the new and powerful appliance to the market.” said Lin Miao, the president of OpenVox, “The modular design of our IX130 is unique in the IPPBX market and it provides full flexibility to all our users by simply installing different telephony interface modules. The punch point is that we have succeeded in reducing the total cost of IX130 and it will come with an eye-catching price.”

The IPPBX supports comprehensive protocol processing, including SIP, H.323 protocols in IP side and BRI, PRI, SS7, GSM and some protocols transcoding in CPE. Taking the full advantages of open source platform, the IX130 will be preloaded with Elastix®, PBX in a FlashTM or trixbox® IPPBX software. A ready-to-work IPPBX as required will be delivered to your doorway momentary.

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elastix · IPPBX · IX130 · OpenVox · pbx in a flash · trixbox

Never thinking of that 16 span T1/E1 asterisk card will come out to the market? Sangoma makes it happen. Even though the official release has not come out yet, the related product info is on the webiste now. Check it out with the following details:

The A116 is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line – using high performance PCI Express interface, providing superior performance in critical systems all over the world. The A116 supports up to 32.8 Mbps of full duplex data throughout 480 voice calls using 16 T1/E1/J1 spans.

With Sangoma boards, you can take advantage of hardware and software improvements, as soon as they become available. The A116, like all boards in Sangoma’s AFT family, is field-upgradable with crash-proof firmware.

Technical Spec
*Sixteen T1/E1 ports with optimum PCI–Express interface for high performance voice and data applications
*Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other ports so both T1 and E1 are supported simultaneously
*Support for Asterisk® and FreeSwitch®
*Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning
*Dimensions: Full height by half length (107mm x 176mm)
*Connector: Dense 68 pin SCSI type interface
*Intelligent hardware: Downloadable FPGA programming with multiple operating modes; add new features related to voice and/or data when they become available
*Line decoding: HDB3, AMI, B8ZS
*Framing: CRC-4, Non CRC4, ESF, SF, D4, J1 (Japan)
*PCI Express Bus Version: 1.1
*Maximum operational power: 5.5 W (1.67 A @ 3.3 V)
*Temperature range: 0 – 50 °C
*Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems
*Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI
*T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications
*Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU
*Uses raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch
*WANPIPE® routing stack is completely independent of TDM voice application for total system reliability
*WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25

Optional DSP Hardware Echo Canceller Daughterboard
*G.168–2002 echo cancellation in hardware
*1024 taps/128 ms tail per channel on all channel densities
*DTMF decoding and tone recognition
*Voice quality enhancement: music protection, acoustic echo control, and adaptive noise reduction
*Does not increase the physical size of the card, and no additional slot is required

Operating Systems
*Windows® 2003, Windows® XP, Windows® Server 2008, Windows® Vista, Windows® 7
*Linux (all versions, releases and distributions from 1.0 up)

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16 T1/E1 · A116 · sangoma

Technology changes, so does our way to hold the official asterisk conference (AsterConference Asia) in China. Hosted by SEC Training & Expo SdnBhd and Co-hosted by VoIP88 Community, the AsterConference Asia 2012 is brought to live with cloud service empowered by Gensee and, it’s a really successful event and a completely different way to share with the community even though with its first presence in the Chinese market.

The open source telephony technologies are widely being used and implemented in the near 2 or 3 years in China. Especially with the localization of the open source community, the telephony market keeps fast growing. Who make it happen? My respect to them: For open source PBX software, there are AsterCC, FreeIris, Elastix (Chinese Version by VoIP88) and so on. For hardware part, we have Digium, OpenVox, Sangoma, Yealink, GrandStream and other vendors. For technical community, there are VoIP88 Community (the biggest Open Source telephony community in China with more than 30,000 registered users), 51asterisk Community, Asterisk-help Community, etc. And thus, bringing AsterConference to China is smart choice.

The AsterConference Asia 2012 was held on 30th and 31st March 2012 in Shenzhen, China. AsterConference is the first ever series of conference in Asia dedicated to Asterisk, the world’s most popular open source IP-PBX telephony software which has greatly revolutionized the telecommunication landscape in the business world. Its power, scalability, feature-rich functions and cost-effectiveness are some main reasons why Asterisk is a clear option for IP-PBX. Later this year, there will be AstriCon in USA and Elastixworld in Spain.

I would like to share this event with you all: videos, pictures and event agenda.

For online videos, please visit Gensee.

 Here are some pictures for the show:

Agenda of the event:

Day One:

09.00am – 09.30am        Registration                           
09.30am – 10.00am       Opening Ceremony / Welcome
10.00am – 10.35am       Doug Vilim (Sangoma Technologies Inc.)
Asterisk – Revolutionizing the High Volume Call Space
10.35am – 11.10am         Solo (AsterCC)
How To Use Asterisk As A Predictive Dialler Solution 
11.10am – 11.40am         TEA BREAK
11.40am – 12.15pm         Greg Vance (Digium Inc)
Asterisk Open Source… continues to innovate
12.15pm – 12.50pm          Karmy Wang (Product Director of Yealink)
The Worldwide Open Opportunities Comes With Asterisk Development
12.50pm – 01.50pm        LUNCH
01.50pm – 02.25pm        Ken Wu (Grandstream Network Inc.)
Innovative IP Voice & Video Solutions
02.25pm – 03.00pm        L.C. Chen (SITA Malaysia)
03.00pm – 03.30pm       TEA BREAK
03.30pm – 04.05pm         Dengshan Xiong (Synway Information Engineering Co. Ltd.)
The Role of Synway Products in Asterisk Application
04.05pm – 04.45pm         Question & Answer
Lucky Draw & Photo Session

Day Two:
 
09.30am – 09.40am          Welcome Speech
09.40am – 10.20am           James Zhu (Hiastar)
Some Concerns When Implementing Asterisk Cards And VoIP Gateways
10.20am – 10.50am           TEA BREAK
10.50am – 11.30am            Udi Delgoshen (Sangoma Technologies Inc.)
11.30am – 12.10 pm           Fang Qiang (Shenzhen Asterisk Network Technology Ltd.)
Introduction of Open Source Project About Asterisk on MIPs
12.10pm – 01.10 pm          LUNCH
01.10pm – 01.50pm          L.C. Chen (SITA Group of Companies Malaysia)
Asterisk & Other Open Source Applications: Contribution to Building and Property Management Industry
01.50pm – 02.30pm         Zhang Yang (VoIP88 China)
How To Develop Applications Based On Elastix
02.30pm – 03.00pm        Question & Answer, Photo Session
03.00pm – 03.30pm       TEA BREAK
03.30PM                               END

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asia asterisk conference · asterisk · digium · elastix · open source · OpenVox · sangoma · telephony · VoIP · voip88

Well, guess what, I still remember last time when Sangoma and Pika released their full product line gateway products to the community and there should be a 3rd board company doing the same soon. And, it’s Digium. A few months later, Digium released its E1 gateway products and without official press announcement(I think I didn’t see any from their website up to now.) But we can have the related info from elsewhere. Here is a post by Christina Smith from VoIP Supply.

What Is It?
The new Digium Gateways are an alternative to installing Digium TDM cards into your server when bridging the PSTN to a SIP environment, or vice versa.

Harness the power of Digium TDM cards in a standalone appliance with no fans or moving parts to convert PSTN to SIP without using precious server resources.

What Does It Do?
Digium single T1 or dual T1 gateways can be used to bridge the PSTN with an IP PBX, to use an ITSP for SIP trunking with a legacy analog PBX, or to migrate to VoIP slowly by bridging the PSTN with both an analog PBX and an IP PBX.

The Digium gateway is also an ideal appliance for bridging remote sites with a central PBX.

The Digium G100 and Digium G200 gateways offer:

Automatic call type detection (voice, modem, fax) and…
Answer and disconnect supervision
Trunk group support
Dial plan support
Caller ID name and number
Fax and modem support (T.38 and G.711 pass through)
The Digium G100 and G200 gateways also provide pass through support for calls to toll free, local, and emergency service numbers.

Digium SIP Gateways are currently offered in a single T1 deployment (G100) or a dual T1 model (G200) and provide support for multiple SIP endpoints. It includes the 100-240 VAC power supply and has onboard echo cancellation.

Who Is It For?
The Digium G100 and G200 gateways are for simple TDM to IP conversion. They are available at a price point almost half of its competitors. They are standards based and should work with any vendor that is also compliance based.

The G100 will support up to 30 simultaneous SIP Calls and the G200 will support up to 60 simultaneous calls. Although there is a limit to how many SIP to SIP calls each appliance can do simultaneously, there is no limit to how many SIP endpoints can be configured on the appliance.

Because of the simplicity of installation, configuration, and use, the Digium SIP gateway can be used for anybody who is looking for simple TDM to IP conversion, but doesn’t want to install TDM cards into a server.

Who Isn’t It For?
I am sure you may have noticed that the Digium G100 and G200 gateways are quite a bit less expensive than competitor gateways. The main function of the Digium gateway is TDM to SIP conversion.

There are some limitations.

The Digium G100 and G200 are not an edge device. They have no firewall or router capabilities – yet. That said, if you need a gateway that can perform these functions now, you should look at an AudioCodes or an Edgewater gateway.

Stay tuned for the next release of Digium gateways, which will have router and firewall capabilities integrated.

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digium · Digium G100 · Digium G200 · digium gateway · IPPBX · VoIP Gateway · voip supply

Sangoma has announced the addition of several new distribution partners in both the European and Asian markets. Each of these new partners will offer select products from Sangoma’s broad array of advanced communications products, including boards, gateways, transcoding solutions, and other advanced communications technologies to network operators and enterprises.

Among the new distributors joining the Sangoma family are: ProVu, a communications provider located in Huddersfield, England; IT-Logiq, a Paris-based provider of converged communications technologies; Avanzada 7, a Malaga, Spain-based company that offers a wide range of voice and data communications technologies; Alias, a voice and data communications supplier headquartered in Udine, Italy; Focus Telecom, a provider of IP communications solutions located in Caesarea, Israel; and Ben International, a Dubai-based company that offers customers a full range of voice and data communications products.

“We are delighted to add these knowledgeable and highly successful distributors to our ranks,” said Bill Wignall, president and chief executive officer of Sangoma. “As a company with a worldwide presence, Sangoma is continuing to expand a distribution network comprised of experienced partners who share our commitment to deliver superior products, expertise and value to customers. These newest partners share Sangoma’s core beliefs, and will be a tremendous asset to the continued growth of our company.”

Sangoma offers a complete portfolio of advanced communications technologies, including hardware, software, gateways, and other solutions to customers in some 150 countries across the globe. Sangoma’s solutions are used in both proprietary and open source networks by network operators, service providers, enterprises, and mid-size businesses.

“Sangoma has earned an outstanding reputation for technologies that are not only innovative and reliable, but most importantly, serve the tangible business needs of customers,” said Darren Garland, managing director of ProVu. “Our relationship with Sangoma will allow us to offer solutions that will enable customers to improve network performance, and leverage the many emerging IP- and cloud-based solutions that are entering the market. We fully expect that this partnership with benefit us, Sangoma, and above all, our customers.”

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gateway · sangoma · transcoding