Digium Adds PSTN Faxing Support to Open Source Asterisk

April 6, 2009 by Garrett Smith

Fresh off their announcement of revamped, comprehensive support package offerings for open source Asterisk, Digium today has released Fax for Asterisk .

HUNTSVILLE, Ala.ā€”April 6, 2009ā€”DigiumĀ®, Inc., the AsteriskĀ® Company, today announced Fax For Asterisk, a complete, cost-effective platform for the development of fax solutions. The offering provides Asterisk users and integrators a suite of user-friendly applications and a licensed version of the industry-leading fax modem software from Commetrex. To meet the demanding requirements of business users, Fax For Asterisk provides reliable faxing across the Internet and public switched telephone network (PSTN).

Asterisk is the most widely used open source telephony platform. The software is available free of charge and has been downloaded millions of times for use by individual developers and systems integrators creating custom telephony solutions for businesses. Asterisk is also available as the professional-grade and commercially supported Asterisk Business Edition.

ā€œAsterisk users, developers and integrators now have a toolkit allowing them to integrate fax with their phone systems,ā€ said Bill Miller, vice president of product management at Digium. ā€œWith Fax For Asterisk, Digium offers a reliable and fully supported fax solution.ā€

Fax For Asterisk interoperates with standards-compliant fax machines connected to Asterisk 1.4 and 1.6 on x86 Linux systems. It provides low-speed PSTN faxing via DAHDI-compatible telephony interface cards as well as VoIP faxing to T.38-compatible SIP end points and service providers. Fax For Asterisk operates at speeds up to 14.4kbps and supports V.17, V.27 and V.29 fax modems.

Fax For Asterisk is available free of charge from the Digium webstore at http://store.digium.com/ for one concurrent fax session. Multi-session licenses are available for a one-time fee of $38.50 per channel. Fax For Asterisk is available immediately. Fax capabilities for Digiumā€™s Switchvox IP PBX were announced in February of this year and are based on this solution. For more details, visit www.digium.com.

Fax solutions for Asterisk are not new (Hylafax, SpanDSP, etc) but direct support from Digium for faxing is, and this plugs another hole in open source Asterisk making it an even more compelling option that has the Shoretel’s of the world looking in their rearview mirror.

The six month VoIP system ROI

This morning Doug Mohney at FierceVoIP dropped some great observations from last weeks VoiceCon show. The one observation that struck a cord here at VoIP Supply was the six month ROI (Return on Investment) businesses want on the purchase of a VoIP system.

A year ago businesses were happy with an 18 month ROI. With new financial pressures (thanks to the economy) an 18 month ROI is no longer realistic for them.

But is a six month ROI realistic either?

Well, let’s take a look.
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Digium Woos SMBs/Enterprises with Support Packages for Open Source Asterisk

OSS Telephony leader Digium recently fired another shot across the bow of traditional proprietary Tier 1 vendors including Cisco, 3Com and Avaya.

In the enterprise space, one of the traditional arguments against open source has been the lack of available support. There are plenty of VARs out there these days helping enterprises of all different shapes and sizes deploy open IP Comms platforms like Asterisk, but nobody will argue that Asterisk’ support ecosystem is as reliable and robust as the likes of Cisco.

Digium is looking to change the perception that Asterisk can’t compete for enterprise customers in several ways. First, they are heavily promoting their DCAP (Digium Certified Asterisk Professional) curriculum to incubate a future army of open source VoIP technicians. Second, Digium is shoring up their own internal support offerings, and recently announced several tiers of paid support for open source Asterisk ranging from 1-3 year increments.

Components that are covered under the new Digium plans includes Asterisk Open Source 1.4, Asterisk Open Source 1.6, Asterisk GUI, G.729 Codec, HPEC, Lumenvox Speech Recognition for Asterisk and Cepstral Text-to-Speech for Asterisk.

Digium is also promising 24X7 availability, Web Case Management, Remote Troubleshooting, Advanced Hardware Replacement, Scheduled Upgrade Assistance, Configuration Review, Performance Review and AGI Script Application Review.

Google voice and voice based search

Ever since Google relaunched GrandCentral as Google voice a few weeks ago I’ve been thinking about what Google voice’s true purpose and future might entail. Many have over-hyped the launch as the death of hardware based phone systems, while others have ho-hummed Google’s efforts as half-baked.

Personally I think that Google voice is merely a feature of greater service offering Google is trying to put together. But what that greater service offering might be is still to be determined.

A hint as to what Google may looking to do with Google voice was uncovered in a BBC story late last week. The story, entitled “Google see voice search as corespeaks to Google’s ambition to dominate the mobile search space through voice activated search.

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Tech Tip: Converting a Cisco IP Phone from SCCP (Skinny) to SIP Firmware

April 3, 2009 by Garrett Smith

Cisco IP Phones are amongst the most popular desktop IP phones out there. By default, Cisco ship their phones from the factory pre-loaded with their proprietary SCCP protocol firmware (also commonly referred to as “skinny”).

If you are running Asterisk, Trixbox, Switchvox or any other standards-based SIP platform or hosted service, you’ll need to migrate your Cisco phone(s) from their native SCCP (skinny) load to SIP in order to use them. While this is not a particularly difficult procedure, it can be frustrating for those who have never attempted the process.

For the purposes of this exercise, we’re using a Cisco CP-7960G. The process may be slightly different depending upon the specific model of Cisco IP phone you are working with.

Cisco 7940/7960 IP phones can support either the Skinny Call Control Protocol (SCCP), Session Initiation Protocol (SIP), or the Media Gateway Control Protocol (MGCP), but not more than one simultaneously. This is possible because they load different firmware versions on bootup. This functionality is transparent to the end user, and you enable it through changes to the basic textāˆ’based configuration files that the phones download from a Trivial File Transfer Protocol (TFTP) server.

First, a few prerequisites:

A – You’ll need a CCO login for Cisco.com in order to obtain the latest SIP firmware. The easiest way to obtain a CCO login is to purchase a Smartnet maintenance contract for your Cisco IP phone from an authorized Cisco reseller. Once you have a registered Smartnet, you can obtain CCO login credentials and access the firmware downloads section of Cisco’s website. Expect to pay $8-$15 for a Smartnet contract.

B – You should have a comfort level with basic networking concepts and TFTP setup/administration. (more…)

VoIP conference phone showcase

Note: See more updated blog post here: Best Conference Phones for Different Room Sizes (Infographic)
No so long ago you didn’t have much of an option when it came to an IP based conference phone. You either got a Polycom or you IP enabled an analog conference phone with a VoIP adapter.

Over the last few months this has changed. A number of new IP conference phones have hit the market, much to the delight of many of you.

But with an influx of choice comes a bit of added complexity in purchasing the right IP conference phone. So in order to help you make better purchasing decisions, here’s a quick round-up of the top IP conference phones on the market today.
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Snom Rounds Out Product Suite with MeetingPoint IP Conference Phone

April 2, 2009 by Garrett Smith

German SIP experts Snom are becoming more and more well known for their excellent line of desktop IP phones. One gap in their product line that has recently been filled is desktop conferencing, with the release of their MeetingPoint IP Conference Phone.

MeetingPoint is a SIP compliant, Wideband Voice capable desktop IP conference phone. Similar to Polycom’s “HDVoice”, Snom calls their HD equivalent technology “OmniSoundĀ®” Full duplex broadband sound technology.

The Snom MeetingPoint provides management of up to 4 external participants and call recording range up to 30 mĀ² room area or 10 participants.

The Snom MeetingPoint is expected to ship in April 2009, and is currently available for pre-order on VoIPSupply.com.

Polycom Announces CX5000 (Formerly Microsoft RoundTable)

You may have heard of Microsoft’s mysterious RoundTable voice and video collaboration device, which was announced quite awhile ago but has never seemed to actually materialize in the channel. It’s no secret that hardware manufacturing is not Microsoft’s strong suit…..thankfully this product concept is finally seeing the light of day thanks to Polycom….a company that does know a thing or two about voice, video and hardware manufacturing.

Polycom CX5000

When used in conjunction with Microsoft Office Live Meeting 2007 or Microsoft Office Communications Server 2007, the Polycom CX5000 delivers a unique, engaging 360-degree group video experience, bringing video, voice, and content together into one seamless interactive session.

The Polycom CX5000 features simple USB plug-and-play operation that require little or no training to use. The product is intended to enhance team collaboration and enable faster and more effective decision making, improved interaction and lower costs. Advanced technology automatically changes the camera view so that the active speaker can always be identified, allowing participants to easily track the flow of conversation. The Polycom CX5000 allows you to fully engage all participants by providing a 360Ā° panoramic view of the conference room when used with Microsoft Live Meeting 2007. In a Live Meeting 2007 session, both video views and the shared content are all seamlessly integrated on one screen. The CX5000 also allows you to record meetings for playback with synchronized voice, video and content, using Live Meeting 2007.

VoIP equipment sales soared in Q4

It’s still too early to tell how the VoIP equipment market performed this past quarter, but iLocus says the VoIP equipment market grew substantially in Q4 2008.

According to their latest research which utilizes data from carriers, VoIP subscriber lines equipment saw quarter-over-quarter growth of 66%. That’s a hefty number in the face of economic fear, uncertainty and doubt.

These numbers don’t speak to the total VoIP equipment market, though one can imagine that if carriers are building out their infrastructures and new lines are being added, the market for premise based VoIP equipment grew in stride.

This is something that we’ve notice here at VoIP Supply. While the many an industry has struggled as has the VoIP industry, there is still strong interest and demand for VoIP system equipment.

And don’t be surprised if iLocus’ Q1 report shows more of the same. From what I’ve seen and heard Q1 wasn’t too bad to most in the VoIP equipment space.

Cisco SPA525G Multimedia IP Phone NOW SHIPPING!

April 1, 2009 by Garrett Smith

The long awaited Cisco SPA525G phone is now in stock and shipping at VoIPSupply.com!

The Cisco SPA525G is compatible with popular SIP based platforms including Asterisk and Trixbox, and is also supported on Cisco’s UC520.

We’ve lowered our price to $299.95, this phone has everything:

    • Support for up to 5 Lines
    • Dual Ethernet with 802.3af PoE Support
    • High Resolution Color LCD Display
    • Integrated WiFi 802.11X Connectivity
    • USB 2.0 host port for connecting a USB memory device to play MP3 music files
    • AUX port (to attach a SPA932 attendant console)
    • Bluetooth capability for headset support
    • 2.5mm stereo earphone jack for wired headset
    • Kensington security slot support
    • Integrated web/XML applications (Weather, News, Sports, etc)
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