Tom Keating : VoIP & Gadgets Blog
Tom Keating
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SIP

New Nimbuzz VoIP app for the iPhone and iPod touch

March 19, 2009

Nimbuzz just released their new iPhone version of Nimbuzz which also supports 3G VoIP "dial up" calling and can turn the iPod touch into an iPhone. The old version was just released into the Apple iTunes store in November, so Nimbuzz is cranking out new version pretty quickly!

The new version features a full dial-pad, and the ability to make VoIP calls to PSTN numbers using SkypeOut, as well as via their 10 VoIP partners including Gizmo5, Vyke, sipgate and A1 by leveraging SIP. You can now add individual buddies from AIM, Google Talk, Windows Live Messenger (MSN), MySpace, Yahoo!, and Nimbuzz.

If Wi-Fi is unavailable you can make VoIP calls to Nimbuzz buddies using what Nimbuzz calls "Dial-Up VoIP", which is available in over 50 countries.

Dial-Up VoIP simply means that Nimbuzz dials a local access number that your iPhone dials and then Nimbuzz's VoIP servers terminate the call. Jajah, and others have this feature as well.





Dotcom-Monitor announces new SIP Monitoring tool

March 18, 2009

Today, Dotcom-Monitor announced a new SIP monitoring tool to add to its portfolio of external monitoring services. It's similar to other web-based Monitoring-as-a-Service (MaaS) services which monitor the uptime of web servers and notify when a problem occurs. In this case, Dotcom-Monitor's SIP Monitoring service monitors on-premise or hosted IP-PBXs.

How's it work? Dotcom-Monitor's SIP monitoring service makes live intermittent SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity.

How to make OCS 2007 R2 non-RFC 3966-compliant using RemovePlusFromRequestURI

March 16, 2009

Office Communications Server 2007 Mediation Server uses a plus sign (+) to prefix E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls. Unfortunately, some IP-PBXs don't comply with RFC 3966 and do not accept numbers that are prefixed with a plus sign (+).

As the UCSpotting blog points out:

To make sure that OCS 2007 operates correctly with non-RFC 3966-compliant PBXs, Microsoft released an update for Mediation Server (R1), which is described in KB articles 952780 and 952785. After installing the update, it's necessary to create a configuration file - MediationServerSvc.exe.config - with the following content:

"1.0" encoding="utf-8" ?> <configuration> <appSettings> <add key="RemovePlusFromRequestURI" value="Yes" /> </appSettings> </configuration>

In OCS 2007 R2, Microsoft changed this slightly negating the need for the above configuration file.




Microsoft OCS 2010 Will Finally Eliminate the PBX

March 16, 2009

Well, Microsoft has let the cat out of the bag and leaked word that Microsoft OCS 2010 will "remove the need for PBX equipment within your organization". I'm certainly not surprised. Let's flash back to last year where I wrote and article titled Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX. In it I wrote:
"Office Communications Server 2007 R2, debuting just one year after the Microsoft unified communications launch, highlights the pace of innovation that is possible with software," said Stephen Elop, president of the Microsoft Business Division at Microsoft.

Windows Server 2008 RDS Does VoIP

March 11, 2009


Terminal Services allows you to remotely run applications as well as perform remote administrative duties on servers. It has allowed remote audio to be streamed over IP from the remote computer to your local computer (audio redirection) but has never allowed the microphone or line-in port to be redirected. If Microsoft did, you could do VoIP. Of course, you'd have to redirect from the local PC to the remote server and not the other way around.

Polycom VVX 1500 Media Phone Game Changer?

March 9, 2009

Today, Polycom has launched the Polycom VVX 1500 touch-screen business media phone, a new VoIP phone that combines IP telephony with business-class video and the ability to integrate with business applications. Recently, Verizon make a big splash with their consumer-class Verizon Hub, a multimedia phone that combines VoIP, Internet access, color screen, video streaming, and more. One could easily make the case that the Polycom VVX 1500 is the "business-class" version of the consumer-oriented Verizon Hub phone.

Although there are many similar features and both could be classified as "media phones", the Verizon Hub does not do video conferencing, since it does not have an embedded camera. The Polycom VVX 1500 on the other hand does have a video camera embedded (2-megapixel) and is therefore more suited to video conferencing, which is more prevalent in the business world any way.

The Polycom VVX 1500 combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone along with Polycom HD Voice (wideband telephony) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content.



SmartSIP Launches for OCS 2007 R2 Enabling Any SIP Phone & Any SIP Trunking Service Provider

March 4, 2009

OCS 2007 R2 won't replace your PBX just yet. However, their latest R2 version adds the ability to do direct SIP trunking, thus bypassing the need for an IP-PBX.

One drawback however is that Microsoft only supports direct SIP trunking with two providers, namely Global Crossing and Sprint. Well that's pretty lame, considering their are dozens of decent SIP trunking service providers and probably hundreds across the entire world.
Fortunately, Mike Stacy an OCS 2007 guru, over at Evangelyze Communications has some products that enhance OCS 2007 R2 functionality. One such product is SmartSIP which launches tomorrow.


SpinVox Transcribes Skype Voicemail & VoiceScribe does the same for Asterisk

March 3, 2009


Skype
users can now have their voicemails converted into text via SpinVox. Today, SpinVox announced that your Skype voicemails transcribed and sent to you via SMS for €0.20/£0.17/25 cents plus the cost of the SMS. SimulScribe, now PhoneTag, is a similar service, that Rich Tehrani uses regularly. GotVoice is yet another one.

But how about another cool TTS app that is currently 'free' and works with the popular open source Asterisk platform? Weavver's VoiceScribe is a beta web-service for Asterisk that converts your voicemail to text and delivers them to you via e-mail.

SHSU Switches Back to Cisco CallManager from Asterisk

February 27, 2009

In 2006, I came across a Network World article, which espoused the fact that Sam Houston State University (SHSU) had switched from the Cisco CallManager IP-PBX to open source Asterisk. I wrote about this news since 6,000 students and faculty were moved off Cisco to the open source Asterisk IP-PBX, which was great news for the open source Asterisk community. This deployment demonstrated that Asterisk could scale and put to rest one of the main complaints against Asterisk.

Well, 3 years have passed, and according to this thread written by Jason Fuermann, who is responsible for SHSU's IP phone system, SHSU has switched back to Cisco from Asterisk. Say what?





EZCallerID.com Hosted CNAM for Enhanced Caller-ID on any IP-PBX Launches

February 25, 2009


EZ Call, Inc. today announced the launch of EZCallerID.com, a new service that provides enhanced Caller ID, also known as CNAM, for VoIP calls. The hosted CNAM service gives you not just the phone number, but the name of the person calling.

Most SIP trunking providers do not provide the caller's name with Caller ID on inbound calls. EZCallerID.com solves this issue by simply having you route your inbound calls to their server. They insert the caller's name and send the call back to your IP-PBX.

How's it work?




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