Digium Releases Switchvox 4.0 (Finally!)

April 8, 2009 by Garrett Smith

Switchvox SMB release 4.0 shipped yesterday….we upgraded our SMB plant here last night without any real hiccups. With the release of SMB 4.0, a great IP communications platform just got better in many ways.

1 – Switchvox SMB now supports “Presence” across multiple sites (server instances). This was an issue with our own deployment in the past, and it is great now that you can unify multiple, geographically dispersed Switchvox SMB systems with a “global” Switchboard, showing you the status of all users and allowing you to control calls across multiple sites.

2 – Fax! Switchvox SMB 4.0 now supports outbound fax and fax-to-email. I have not had a chance to test this personally but am looking forward to accidentally knocking our finicky analog fax machine off the desk soon.

3 – More API goodness! Digium/Switchvox are wisely investing dev cycles in their API Call Control, unlocking more and more flexibility and integration abilities.

4 – Jabber/Chat capabilities with Switchboard. Perhaps soon I can ditch Windows Live Messenger and AIM in favor of unified IM/Chat within Switchboard.

5 – More / Better Microsoft Outlook integration!

6 – Improved system diagnostics and schedule reports.

There’s more to love in Switchvox SMB release 4.0 beyond the features I’ve mentioned here. If you are considering Switchvox or have any questions about this product, feel free to contact one of our product specialists at 800.398.8647!

Polycom Releases New SIP 3.1.2 Firmware for SoundPoint IP Phones

April 7, 2009 by Garrett Smith

Polycom has just released new SIP firmware version 3.1.2, and it can be found here. This release is identical to the SIP 3.1.2 release with the addition of software and configuration parameters applicable to the VVX 1500 phone.

Here are a few of the highlights:

SIP 3.1.2 includes GA level support for SoundStation IP 7000 integration with Polycom HDX video systems. This feature requires that the SoundStation IP 7000 is running BootROM 4.1.2 or newer software.
The SoundPoint IP/SoundStation IP XML API feature is now formally supported in this release. This feature was designated as a Beta in the SIP 3.1.0 and SIP 3.1.1 releases.
SoundPoint IP 550, 560, 650 and 670 products require BootROM 4.1.0 or newer in order to load SIP firmware release 3.1.2

If you are considering upgrading, make sure you have the correct bootrom, depending upon the Polycom SoundPoint model you own, as follows:

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Linksys WIP310-G1 Wireless Phone Back in Stock!

by Garrett Smith

Just a heads up we’ve received a few hundred Linksys WIP310-G1 WiFi SIP Handsets into stock and they are going quickly. If you have been waiting for one of these, now’s your chance!

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.—April 6, 2009—Digium®, Inc., the Asterisk® 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.

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

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 Digium’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

by Garrett Smith

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

by Garrett Smith

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.

More from: Asterisk Garrett Smith digium

Google voice and voice based search

by Garrett Smith

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 core” speaks 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 text−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

by Garrett Smith

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 “OmniSound®” Full duplex broadband sound technology.

The Snom MeetingPoint provides management of up to 4 external participants and call recording range up to 30 m² 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.

Digium Releases Switchvox 4.0 (Finally!)

Polycom Releases New SIP 3.1.2 Firmware for SoundPoint IP Phones

Linksys WIP310-G1 Wireless Phone Back in Stock!

Digium Adds PSTN Faxing Support to Open Source Asterisk

The six month VoIP system ROI

Digium Woos SMBs/Enterprises with Support Packages for Open Source Asterisk

Google voice and voice based search

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

VoIP conference phone showcase

Snom Rounds Out Product Suite with MeetingPoint IP Conference Phone

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