Tom Keating : VoIP & Gadgets Blog
Tom Keating
CTO
| VoIP & Gadgets blog - Latest news in VoIP & gadgets, wireless, mobile phones, reviews, & opinions

SIP

Google Voice Meet Asterisk

March 23, 2009

Nerd Vittles has another cool Asterisk recipe that combines Google Voice, voicemail transcription (via Google Voice), free calling, and of course Asterisk. Nerd does some packet sniffing and determines that Google Voice, powered by Grandcentral, is using SIP. What's most interesting is that Nerd determine that your SIP connection and your Google Voice phone bill is only protected by a 4-digit PIN. Yikes!

Skype For SIP Marries Skype and IP-PBXs

March 23, 2009

Today, Skype announced it is enabling SIP-based IP-PBX to connect to the Skype network, which will allow low-cost SkypeOut calling, receiving calls from Skype users, and receiving calls from regular PSTN phone lines. Outbound calls from IP-PBX SIP handsets to Skype phones is not part of this news announcement. Skype commented it is too difficult to dial Skype usernames from a desktop handset.

Features:
  • Receive and manage inbound calls from the 405 million Skype users worldwide on SIP-enabled PBX systems, connecting the company website to the PBX system using Skype click-to-call buttons
  • Place calls via Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX, saving your business money with Skype's low rates
  • Purchase Skype online numbers to receive calls to the corporate PBX from landlines or mobile phones
  • Manage Skype calls using your existing hardware and system applications such as call routing, conferencing, phone menus, voicemail and call recording and logging - no additional downloads or training are required

Skype For SIP is perfectly suited to businesses that already have IP-PBXs and want to connect to Skype's network which offers low-cost calling. Skype for SIP is being launched as a closed beta program, but you can register and try to be part of the beta.



Luca's Top 30 VoIP Leaders on Twitter list

March 20, 2009


Luca Filigheddu has a Top 30 VoIP Leaders On Twitter post worth checking out. TMC's Rich Tehrani and I are on the list. Honestly, I haven't seriously started using twitter until the beginning of this month, so I'm just ramping up my followers and who I follow. So I'm grateful to Luca for still considering me for the list considering my twitter nascency.

Many good people worth following are on Luca's list which is in alphabetical order. Make sure to check out the list and follow them if you are interested in VoIP, including such topics as Skype, Vonage, SIP, Packet8, Asterisk, FCC regulation on VoIP, etc.


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.


Featured Events