Asterisk

Asterisk Linux Based IP-PBX

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.

Posted in Asterisk, SIP, Skype, Unified Communications, VoIP

MFC/R2 support added to Asterisk 1.6.2

March 17, 2009

Posted in Asterisk, VoIP

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 â¬0.20/Â£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.

Posted in Asterisk, SIP, Skype, TMCnet, Unified Communications, VoIP

PIKA WARP Appliance Adds BRI Support

March 3, 2009

PIKA Technologies announced today the release of a BRI expansion module for the PIKA WARP Appliance. The PIKA WARP Appliance is a very flexible hardware telecom appliance that can run various flavors of Asterisk, including native Asterisk, Schmooze, trixbox CE, and others. They even support FreePBX, the popular front-end GUI for Asterisk. They support FreeSwitch as well.

PIKA's BRI module supports two ports and four channels, allowing the WARP Appliance to reach a total port density of four ports and eight channels when two BRI modules are installed.

Posted in Asterisk, TMCnet, VoIP

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?

Posted in Asterisk, Cisco, SIP, TMCnet, VoIP

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?

Posted in Asterisk, SIP, VoIP

Build your own SIP-to-Skype gateway using Asterisk

February 17, 2009

While we wait for Digium's official SIP-to-Skype gateway, Nerd Vittles today informed me about his very cool recipe that you can use today to build your own free SIP-to-Skype gateway enabling you to use your SIP-based desktop phones connected to Asterisk to make Skype inbound/outbound calls.

Part of the recipe uses SipToSis - SIP to Skype Gateway Bridge Proxy. SipToSis is a piece of software which Nerd Vittles points out "forms the lynchpin of Gizmo's offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make."

Nerd Vittles explains in his tutorial:

When we're finished, you'll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You'll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server.

In summary, with this tutorial you'll be able to dial Skype users, as well as receive incoming calls from any Skype user!

Posted in Asterisk, SIP, Skype, TMCnet, VoIP

Bootable Flash Drive creates 15-Minute Turnkey Asterisk Installs on Atom-based processors

February 4, 2009

How'd you like a bootable USB flash drive which can create turnkey, full-featured Asterisk PBX systems in 15 minutes or less? Nerd Vittles has a recipe specially designed for the new Atom-based motherboards found in most netbooks. Nerd says it also works just fine with Dell's PowerEdge T100 servers.

He writes how this is perfect for on-site demos:

For those that demo systems for a living, no one will touch this presentation. Just show up at a customer site with a $300 Acer Aspire One NetBook and an Aastra 57i business phone.

Posted in Asterisk, SIP, TMCnet, Unified Communications, VoIP

ContactQ Enhances Asterisk's Call Center Functionality

February 4, 2009

ContactQ is a new call center application server created by Braxtel Communications designed to run on Asterisk that brings advanced call center functionality to the Asterisk platform. Their aim is to handle any sort of contact method and put it into their advanced multi-media queue. For instance, they plan on queuing video calls, text messages, web callbacks, and of course regular calls. ContactQ is a fully featured multi-media skills based routing ACD.

Posted in Asterisk, Call Center and CRM, ITEXPO, Unified Communications, VoIP

Switchvox SMB 4.0 launches at ITEXPO

February 2, 2009

I just got into Miami for ITEXPO headed to the convention center around 12pm and I've already learned some exciting news launching at the show. Digium has released Switchvox SMB 4.0 with some new Unified Communications features. Matthew Nickasch, a writer for Considering Convergence on NetworkWorld.com covers the news extensively in his ITExpo East Kicks Off with Big Switchvox News article.

Off the bat I noticed the video phone support with support for H.263 and H.264 codecs. Another interesting new feature is XMPP support for a private chat server.

Posted in Asterisk, ITEXPO, Polycom, TMCnet, VoIP

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