Techday Asia - Enterprise Asterisks Conference @Kuala Lumpur Organised by Intuit and Strateq

29 Feb 2012, Sime Darby Convention Center from 9.00 am until 5.00 pm i attended Techday Asia - Enterprise Asterisks Conference. I learned new term 'asteriskfication' and 'asteriskfy' from sunjay ws during his presentation titled "How and why to integrate Asterisk with your PBX" . You can't find this term in Oxford dictionary. The word asterisk is the special noun, asteriskfication is a noun refering to "the integration between PBX and asterisk to activate Voice Over Internet Protocol (VOiP) using open platform" and 'asteriskfy' is a verb refering to 'the action of asteriskfication'. By doing this you can save your corporate phone bill more than 50 percent.

Small businesses, enterprises and even governments use Asterisk-based technologies to enable feature-rich voice communications over a Web connection. But not all current and prospective Asterisk (News - Alert) users know the true power of the open source software. The capabilities provided by Asterisk stretch well beyond traditional voice.

The software also enables next-gen communication functionality like text-to-speak, automatic speech recognition and voicemail/email integration, which allows users to unify their email and voicemail at no charge and natively store voicemails in IMAP.

In addition, Asterisk users can leverage the power of Google services to place and receive free calls using Google (News - Alert) Voice and GTalk. Asterisk servers also sport a calendaring API that can be integrated with CalDAV, Exchange and iCalendar.

On the business side, Asterisk can connect users to directories and authenticate them against an LDAP server, Lotus Domino Directory, Apple OpenDirectory, or Microsoft (News - Alert) ActiveDirectory. Servers can also be integrated with compatible databases for user management, enabling the logging of calls and authentication of users.

Want to have a little more fun with a phone system? Asterisk has been used on some occasions by IT administrators to make a game out of their call queues. Hold music can be replaced by interactive trivia games that keep callers from pulling their hair out from listening to Michael Bolton or a silent line. Even more practical, administrators can use Asterisk to set up a system where callers can hang up and get a call back when a customer service representative is freed up.

Other fun differentiating options offered by Asterisk include PITCH_SHIFT function, which alters the pitch of a caller's voice, and JACK, a new offering that can modify audio in fun ways. Possibly the most underrated benefit of Asterisk-based solutions is the high-quality audio that they provide. The latest version of the software can provide CD or even Blu-ray quality sound.

Asterisk vs. Cisco Unified Communications

BY DJ MONROE

Over my 12 years in the Telecommunications Industry I have worked with a variety of phone systems including Avaya, Aspect, Nortel, Cisco and Asterisk. I am often asked how Asterisk compares to other traditional phone systems. These days I am most often asked to compare Asterisk with Cisco. Many times I am told by a prospect that Cisco can do things that Asterisk cannot, or that Asterisk is not as reliable as Cisco. Allow me to set the record straight regarding the two systems.

Asterisk vs. Cisco, here are some points where we can differentiate. Since I have administered both systems, I can speak with authority on this subject.

1. Cost – Even if Cisco undercuts their cost upfront, they will make it up on the backend. If you buy a Cisco system you will pay a license for every extension on the system, the phones are more expensive, and you have to buy Microsoft exchange licenses for each voicemail box on the Unity voicemail system. On top of this, you will pay for annual support from a Cisco partner who will charge 20% - 30% of the total cost of the system yearly.

2. Features – In some areas the Cisco phone system really excels. The distributive architecture of the system is quite nice. All in all it is a great phone system. However, out of the box if you compare feature for feature Asterisk can do much more. In addition, due to the openness of its architecture you can make Asterisk do pretty much anything that you want.

3. Voicemail Systems – The voicemail system that comes free with Asterisk is 100% better than the Unity voicemail system that Cisco uses. Unity relies on a Microsoft Exchange mail system to manage voicemails. This is a needlessly complex design that does not provide any enhancements to the overall features of the voicemail system. In addition, on the Cisco system voicemail administration is separate from user/extension administration. Therefore, in addition to logging into the Cisco Call Manager to manage the user and extension, the administrator has to log on to a completely separate system to administer voicemail. With Asterisk, combined with our device management software, the User, Extension, Voicemail and device configuration are all managed from one screen.

4. Phones -- Cisco makes a great phone, however in my opinion Polycom makes the best devices currently on the market, and they are priced much lower than equivalent Cisco phones. Furthermore, Asterisk allows you to choose the phones you want to use with your system. Good luck attaching and managing anything to Cisco's Call Manager that does not have a Cisco logo emblazoned on it.

5. System Integrity – Like any application there are good ways to deploy Asterisk and there are bad ways to deploy Asterisk. If you ask around organizations that have Asterisk based phone systems you will find an array of experiences. Many of these organizations will tell you that their phone system is their biggest nightmare. Still others will tell you that their Asterisk based phone system is the best thing since sliced bread.

In most cases the difference between these organizations is their deployment methods. Many organizations are attracted to Asterisk because of cost. Sometimes this leads those same organizations to cut corners on implementation, phones, interface cards, and server hardware. With Asterisk you get what you pay for. Cutting corners now will cost you in the future. Therefore, hire an experienced integrator to build and support your system, listen to their advice when purchasing phones, and don’t cut corners on equipment costs. PLEASE DON'T EVER BUY CHEAP PHONES! Unless you have a staff member that has performed several Asterisk deployments, don’t do it yourself. This strategy will ensure that you have a quality phone system that rattles and hums as it drives up productivity while enhancing the workplace environment, and you will still come out way ahead on cost over a traditionally branded system.

6. Future Proof Technology – Asterisk is open, freely available, and developed by a community of developers committed to constant improvement of the product. This ensures that the latest enhancements and fixes are available to you without the purchase of new software, licensing, or equipment. If the latest version of Cisco's Call Manager arrives on the market with some fancy new feature, plan on starting from scratch to upgrade your environment. In many cases a Cisco Call Manager upgrade will require you to purchase new software, pay for the same licenses again, and often times buying new rebranded HP servers that have been marked up by Cisco three times their original retail price from ~$4000 to ~$12,000.

Several Cisco shops have dumped Cisco in favor of Asterisk. In many cases they cite cost and features as the reason:

http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-replaces-cisco-callmanager.asp

http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5

Here is another interesting article that discusses why some people integrate Asterisk with Cisco:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration

Needless to say they are both good systems, but at the end of the day open standards, ongoing cost of ownership, and flexibility win in my opinion.

If you are interested in a full list of Asterisk and Cisco features, they can be found here:

http://www.asterisk.org/support/features

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps556/solution_overview_c22-493511.html

Here is a very good post comparing Polycom and Cisco Phones:

http://asterisk.mdaniel.net/?p=11

For more information about BitWare Technologies and our product please visit our website:

http://www.bitwaretech.com

Asterisk Resources

http://www.asterisk.org/ - Latest Source Code

http://www.digium.com/ - Asterisk TDM hardware

http://www.voip-info.org/ - General VoIP How-To Info

http://www.xten.com/ - Softphone

http://www.asterisk-vonage.com/ - Asterisk to Vonage connectivity

http://www.binary-systems.com/ - Asterisk Consulting & Training Services

Skype Ends Support For Open-Source Digium Asterisk VOIP PBX

Skype has terminated its partnership with Digium, effectively killing Skype for Asterisk, which integrated the VOIP service with the open-source telephony platform.

Two weeks after Microsoft said it would acquire Skype, the voice over IP company has begun cutting its ties with the open-source world. Asterisk was the first partner cut loose.

Skype decided not to renew its agreement with Digium, which allowed Asterisk, the open-source telephony system, to be integrated with the Skype service, Digium said May 25 in a letter to users. Digium is behind most of the work on Asterisk and sells commercial products based on the platform. Skype for Asterisk uses some proprietary code from Digium to connect the two products.

"It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software,” according to Digium’s letter.

Skype for Asterisk sales and activations will end on July 26, but Skype has promised to continue supporting and maintaining the software for two more years. Skype may extend this time period “at their discretion.”

Many businesses and governments around the world rely on Asterisk for its free and flexible PBX to power their VOIP deployments. The integration with Skype gave access to low-cost voice and video calls without complex integration. After July 26, new Asterisk users will not be able to connect to the Skype network.

Skype may be moving away from Asterisk because Microsoft is expected to launch a Microsoft-hosted version of its Lync unified communications server this summer. Asterisk competes directly with Lync.

Digium may have seen this one coming, as the CEO predicted shortly after the $8.53 billion deal was announced on May 13 that Microsoft may “wall-off” Skype from competing products. Microsoft’s tendency toward “notoriously proprietary tactics” will slow the development of Skype as a business tool, Danny Windham, Digium’s CEO, wrote on the company blog.

“Microsoft plus Skype equals Microsoft,” Windham wrote.

While it may be easy to pin down Skype’s decision as yet another example of Microsoft trying to shut down the open-source community, Tim Panton, a Skype developer, pointed out on his blog that Skype for Asterisk has been dying slowly for a while because of issues with scalability and maintenance. Skype had “hobbled” the product with a number of license restrictions and the company delayed development, according to Panton.

“Skype probably never envisaged renewing, so when it came due, they pulled the plug,” Panton said.

“I do love a good conspiracy, and it would be great to pin this on Microsoft,” wrote Dave Michels, president of Verge1 Consulting, specializing in PBX strategies. However, Michels noted that it was unlikely Microsoft was calling the shots when the deal hadn’t even closed yet.

Michels also pointed out that while Asterisk is open source, Skype is not, so claiming Microsoft will ruin Skype because of its anti-open-source stance is “hypocritical jabberwocky.” Skype uses its own clients, its own codecs, its own signaling and its own firmware licensed to hardware partners. It does not interface with any other networks or equipment other than basic voice services. “If anything, Skype might teach MS a thing or two about being proprietary,” Michels said.

A Gartner analyst agreed that Skype’s decision had nothing to do with Microsoft. This is a sign Skype will open up its service to other telephony platforms via Skype Connect, Steve Blood, research vice president and agenda manager at Gartner, told eWEEK. While Skype for Asterisk was a bit deeper than what Skype Connect (formerly Skype for SIP) offers for other telephony platforms, it's a "stronger business proposition" for Skype to offer more customers Connect than to support a proprietary product for a specific vendor, according to Blood.

"I don't think Skype for Asterisk was compelling enough, nor did it generate enough money for Skype to continue to support it," Blood said.

Skype Connect currently works with telephony systems for Avaya, ShoreTel, and Cisco, among others. Digium will be validating Skype Connect next month, according to Blood, so Asterisk customers will continue to have some Skype support.

(Source - http://www.eweek.com)

Sangoma Sponsors Asterisk Community Portal - First of Three Educational Online Community Portals is Launched

TORONTO, ONTARIO--(Marketwire - 05/13/11) - Sangoma Technologies Corporation (TSX-V:STC - News) , a leading provider of hardware and software components that enable or enhance IP Communications Systems for both voice and data, today announced the launch of the first of a series of Global Online Communities on TMCnet. TMCnet is the world's most-read B2B website covering communications technology.

Recognizing the need to continue the education of decision-makers, influencers, developers and customers in the world of communications and technology, Sangoma has partnered with TMC to sponsor and update three Global Online Communities. The first of these, the Asterisk Global Online Community goes live today.

In the coming weeks, Sangoma and TMC will partner again on an IP PBX Global Online Community and an IP Telephony Global Online Community.

"We realized that while these three topics are all closely related, there are real distinctions in the decision making and implementation issues addressed by each of the technologies and markets that are addressed by these communities," said Jeff Dworkin, director of marketing for Sangoma. "They are all in states of rapid transition and we want to help as many companies as possible navigate these ongoing transitions."

The Asterisk Community will serve to introduce new players to Open Source Telephony and advance innovative developments around that technology.

The IP PBX Community will help those who are just now moving their Businesses and Enterprises from TDM-based technologies to an all IP-based infrastructure.

The IP Telephony Community will serve as a resource for Developers, SMBs, Enterprises and Carriers who are looking for the most innovate and up-to-date information and solutions in this space.

About Sangoma Technologies Corporation

Sangoma is a leading provider of hardware and software components that enable or en-hance IP Communications Systems for both telecom and datacom applications. Sango-ma's data boards, voice boards, gateways and connectivity software are used in leading PBX, IVR, contact center and data-communication applications worldwide. The product line in-cludes both hardware and software compo-nents that offer a comprehensive toolset for deploying cost-effective, powerful, and flex-ible communication solutions.

Founded in 1984, Sangoma Technologies Corporation is publicly traded on the TSX Venture Exchange (TSX-V:STC - News). Additional information on Sangoma can be found at: www.sangoma.com.

The TSX Venture Exchange does not accept responsibility for the adequacy or accuracy of this release.

(Sources - http://finance.yahoo.com)

Asterisk powers

Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism. A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers).

For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks. In addition, single to quad port analog FXO and FXS cards are available and are po***r for small installations. Other vendors' cards can be used for BRI (ISDN2) or quad- and octo- port BRI based upon CAPI compatible cards or HFC chipset cards.

Lastly, standalone devices are available to do a wide range of tasks including providing fxo and fxs ports that simply plug into the LAN and register to Asterisk as an available device.

SIP Phone, IP Phone, USB Phone, USB VoIP Phone, SIP ATA, Skype phone

Huntsville, Ala., and Xiamen, Fujian—[27^th Feb,2010]—Digium®, Inc., the Asterisk® Company, and Yealink, a professional IP voice and video phone designer and manufacturer for broadband networks,today announced they have completed interoperability test in Asterisk Business Edition and Yealink SIP-T2x series. Yealink SIP-T2x series is high performance and affordable SIP telephones that help businesses leverage the increasing benefits of voice over Internet protocol (VoIP) telephone system. Yealink SIP phone provides the high quality audio, a brand range of voice codec, security protection for privacy, and rich telephone features. Yealink enterprise HD IP phones now are compliance-tested by Digium for compatibility.Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software.

The company’s product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems. Resellers, telecom professionals and software developers choose Digium’s products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.“The combination of Digium and Yealink enterprise HD IP phone gives businesses a powerful and yet cost-effective choice in VoIP solution market” commented David Chen, the CEO of Yealink Network, “We believe we are providing the best price-performance benchmark in the industry, Digium and Yealink combination offers a very compelling value to SMB”

About Yealink Network Technology

Yealink Network Technology Ltd. is a professional designer and manufacturer of innovative, affordable, and high quality IP voice and video products for the world-wide broadband telephony market. The company's products such as SIP-T2x series enterprise HD IP Phones, equipped with the TI chipset and TI Voice Engine, offer high definition voice, are fully compatible with the SIP industry standard, field proven with large and rapidly growing deployed base, and have broad interoperability with the major IP-PBX, IMS, NGN, soft-switch and other 3rd party SIP products on the market today. The extraordinary performance of our products and professional R&D team ensure us a strategic partner to Digium. For more information, please visit www.yealink.com

About Digium

Digium®, Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company’s product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom telephony solutions. More information is available at http://www.digium.com.

(Source - SIP Phone, IP Phone, USB Phone, USB VoIP Phone, SIP ATA, Skype phone)