Source: www.voip-news.com
VoIce Over IP Weblog has some suggestions for more efficient messaging. As in, forget the useless mp3 voicemails. Read it here.
Word has it there is an upgrade to SkypePhone coming down the pike with a Wi-Fi aspect. Read about it on VoIP News of the UK.
VoIP Watch also has some details about this new SkypePhone.
Source: snapvoip.blogspot.com
Fall 2007 VON Conference (Booth #1261) – Boston, MA – October 30, 2007: TelcoBridges Inc, announces the arrival of Toolpack API, making telecom application development simple and open.With Toolpack, a high-level, open set of C++ development tools, developers can easily and rapidly create new solutions, or migrate existing telecom applications to TelcoBrigdes proven carrier-grade platform. Developers can now benefit from a telecom architecture that delivers incredibly fast time-to-market and unbeatable cost-efectiveness options never before available to the market.
Toolpack API includes a complete set of pre-developed, ready-to-compile and customizable C++ modules for readily developing telecom systems. The set of development tools includes pre-developed building blocks such as:
- A reference telephony application
- Transparent SIP, SS7, and ISDN call bridging
- IVR, voicemail and web-based system management
- Database interfaces, call routing, and more.
Ready-to-compile C++ modules can be integrated into the application code, dramatically reducing the development cycle and allowing developers to focus their efforts on creating value- added functionality. Plus, the Toolpack API source code is available to developers, providing them complete flexibility to customize their solutions both now and in the future.
The Toolpack API application development environment benefits from an underlying HA and Resource Manager runtime software managing dynamic configurations, hardware resources, redundancy switchovers, and clock synchronization. Toolpack transparently supports SIP, SS7 HA, ISDN, full hardware and software redundancy, as well as TelcoBridges industry leading Tmedia Switch. Now with Toolpack API, and with ExpresSCE+, our GUI-based service creation and delivery software platform, TelcoBridges is delivering the industry’s fastest time-to-market and most cost-effective telecom platform building VoIP or TDM solutions," states Gaetan Campeau, President, Founder and EVP Sales of TelcoBridges. "Today’s announcement is a vital step in executing our plan to transform the communications technologies industry into one that is truly dynamic and open".
Toolpack API is available effective immediately for Linux, Intel/SPARC Solaris and Windows operating systems.
www.telcobridges.com
Source: voipcentral.org
VoIP startup Vonage has settled down a legal dispute with Klausner Technologies and reached out an agreement to use their patents relating to voicemail services. This is the second major settlement of Vonage this week.
Earlier this week, the VoIP pioneer agreed to pay some $80 million to Sprint Nextel as part of the settlement. In return, Sprint Nextel has granted more than 100 VoIP patents to Vonage.
It is nice to learn that Vonage is trying to come out of legal hassles and focus on its core business, which in the long turn will help the company to rebuild its brand image in the market.
Judah Klausner-founded Klausner Technologies sued Vonage last year claiming Vonages voicemail platform infringes its technology and sought $180 million damages.
Klausner Technologies controls nearly 25 patents relating to IP-based voicemail services. The company had earlier allowed AOL to use its voicemail patent.
Vonage is still fighting a legal battle with Verizon Communication. The residential VoIP company was ordered to pay $58 million in damages, plus 5.5 percent royalties on future revenues for the violation of three Verizon patents.
Recently, the U.S. Court of Appeals has vacated the $58 million judgment against the residential VoIP player under the ground that the lower court had wrongly interpreted one of the three patents.
Source: snapvoip.blogspot.com
Digium, the Asterisk® Company, announced at Astricon, a new Hardware module and a Software-based High Performance Echo Cancellation Solution for carrier-class voice networks to improve audio quality and reduce VoIP infrastructure costs.
For Asterisk users that connect to the PSTN, the most common type of echo is hybrid echo - the echo introduced by the impedance mismatch between 2-wire and 4-wire telephone circuits. The echo manifests as a distorted and delayed reflection of the users voice while in conversation with an external party through the PSTN.
The new Digium High Performance Echo Cancellation (HPEC) solutions eliminate the need for the tuning and training sometimes required to control echo conditions in Asterisk-based voice systems. The HPEC is G.168 compliant and "toll-quality" as determined by AT&T’s Voice Quality Assessment Labs using a stringent series of subjective and standardized, objective tests. The Digium HPEC outperformed AT&T’s benchmark lab echo cancellers in all testing categories.
"As Asterisk continues its rapid rate of deployment on a wide range of telephone equipment from vendors around the world, Digium is committed to delivering our customers the absolute best quality possible," said Mark Spencer, CTO of Digium. "Digium’s new high performance echo cancellation solutions once again set the industry standard for open source, carrier-class voice networks."
Asterisk offers a strategic, highly cost-effective approach to voice transport over IP, TDM, switched and Ethernet architectures. Digium’s offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.
Pricing and Availability
Digium’s HPEC hardware module solution retails for $235 USD for up to 32 channels. Digium’s HPEC software is available at no cost to in-warranty Digium analog interface customers, and available for $10 USD per channel to non-Digium customers. Availability is immediate.
Source: voipcentral.org
US-based VoIP giant Vonage has launched an innovative service that automatically transcribes voicemails to text and sends them to users email.
Known as Vonage Text, the new service enables customers to read their voicemails anywhere and anytime using their email accounts on their phone, laptop or mobile device.
With Vonage Text, the customers can track their urgent messages and respond them quickly. They can also search for specific information within e-mail messages. It saves time and makes voicemail service more attractive.
As of now, Vonage has some 2.4 million registered customers. The company will no doubt make all efforts to lure them towards the new service.
Source: snapvoip.blogspot.com
VoIP IP Telephony @ http://snapvoip.blogspot.com
Sys Admins view of SER guts!
The following information is from SER site news report, which in turn plucked from a discussion on a Developers mailing list. It is so important so that I have entirely reproduced the post, for my own reference.
"Consider a more simple SIP proxy like repro. All you can do there is start the damn thing and give it the user data (what would be subscribers, aliases, and parts of the usr_preferences in SER 0.9). Sounds all nice and simple.
Now, as an VoIP operator, my world will be a little bit more complicated. I may have different services that run on separate proxy farms. I may have interesting add-on services like call forwarding, voicemail, IVRs, whatever else product management comes up with. Somewhere in a dark corner, I have some PSTN gateways or, instead, I have an agreement with some telco to do that for me.
If you draw this, you’ll get at least half a dozen boxes with weird connections between. If this doesn’t scare you, start sketching the call flows. You will suddenly find little funny quirks, that of course you can put into C code but if why? SER provides you with the opportunity to solve pretty much all of them in a very simple language.
Better yet: You write your script, you do a test call. If it doesn’t work, you make a trace, you fix your script and try again. No compiling, no packaging, just a restart (BTW, something for the wish list: reloading the config on a SIGHUP). Another trace, another round.
Now we fast forward a bit. Your system is running just fine. But one of your PSTN interconnect partners updates their software and — surprise — all the calls to them fail. Sure, you could use another partner. But your friends in billing will tell yet that their prices for some destinations are just insane. We _really_ have to have that first partner.
Sure, you quickly figure out what the problem is. Sure, you call them and try to explain to the unfortunate fellow on the other end how SIP works and why their stuff isn’t really SIP. Sure, after a while they give in and promise to fix it. But can they do that quickly? Nope. They have to go and talk to whoever delivers their software.
Half a year passes and nothing much happens.
Now, with SER all I need to do is find the route for the specific partner, do the magic with subst() and maybe some other horrible things and voila, it works. Everyone is happy. And should the partner actually ever get their stuff fixed, I can just remove those three lines I had to add.
With repro, things would have been quite different. I have to know enough C++ to actually grok their design or have to have someone doing this. Implementing the three line fix, testing it, producing it easily takes a man-day. With SER I did that in three minutes. Including
the test call.
What it comes down to is, that there is no universal thing. For NATi, there isn’t six funny devices that you find a work around, report to the good folks at iptel, who then add another flag. NAT routers change with every software revision. Old things go away, new things pop up. It is your responsibility as a provider to stay close. That’s what people pay you good money for.
Sure, SER is hard to get into as a beginner. If you want to stay a beginner and don’t care about SIP, use repro. It’ll probably work for you out of the box. If you expect to have to do more, invest the time, learn SIP, learn the ser.cfg. It will pay off later. Everything will be "SER gut" (Sorry, that just had to happen)."
Source: snapvoip.blogspot.com
The Asterisk Appliance is a standalone embedded PBX. Targeted for small to medium businesses (2-50 users), and remote branch offices of larger organizations (2-50 users per site), the Digium Asterisk Appliance will feature the commercially licensed Asterisk Business Edition software and the first Digium-developed Asterisk GUI framework.
The Asterisk Appliance appears to take advantage of the Blackfin DSP’s microcode programmability by implementing echo cancellation, and possibly other telephony functions, in hardware.
The Appliance’s I/O includes eight analog ports (FXS, FXO), a WAN port, four LAN ports, hardware echo cancellation, and a "craft port" for debugging. Expansion is available through a CompactFlash slot suitable for voicemail storage cards or wireless radio peripherals. 
The Asterisk Appliance Developer Kit was designed to allow developers to begin working on solutions based on the Asterisk Appliance before its general release. By developing new business applications using the AADK, kit adopters will have an opportunity to become authorized Asterisk Appliance Partners, qualifying for special programs, pricing and priority availability on production products built on this platform.
Purchase of the AADK includes:
* (1) Asterisk Appliance
o Complete Asterisk server with Asterisk GUI framework
o Up to eight analog telephony ports, configurable via modules
o One 4- port FXS module & two 4-port FXO modules
o Slots for Compact Flash and MMC add-on cards
o 8 MB onboard flash
o 64 MB onboard RAM
o 5 Ethernet ports (4 LAN, WAN)
* Cables for all port types
* IP-430 Polycom phone
* CD with all software
* Documentation and specifications
* How-to manuals
* Digium support details
* Asterisk memorabilia
It is priced at around $4K
Get more info and purchase this at Digium, follow the links.
Links;
Asterisk Appliance Developer Kit
Source: snapvoip.blogspot.com
BUSINESS WIRE News article reports that Sprint has today announced the launch of Sprint Wireless Integration, a product that extends customers’ premises-based PBX features and functionality to their mobile phones. The solution offers business customers additional value and new capabilities by integrating Avaya “Extension to Cellular” capabilities and new Sprint network advancements.
Sprint Wireless Integration features include providing users with one phone number that simultaneously rings both the desk phone and mobile phone, along with one converged enterprise voicemail inbox. It also extends PBX features like conferencing and call forwarding to the mobile phone so users can get all the functionality of their desk phone even while away from the office. For example, mobile users can make intra-company calls by simply dialing the four-digit extension of the person they want to reach, just as they would from the office desk phone – with no access numbers to dial or codes to enter first.
Built within Sprint’s IP Multimedia Subsystem (IMS) architecture, Sprint Wireless Integration is the industry’s first "hosted mobility" solution. “By converging wireline and wireless functionality, Sprint Wireless Integration provides a better overall service – one that is more functional and also makes communication more simple and effective,” said Tony Krueck, vice president of product management and development, Sprint. “This solution is a great example of the promise of Fixed/Mobile Convergence.”
Sprint Wireless Integration provides:
Features
* One phone number with simultaneous ring to both the desk phone and mobile phone (using the existing desk phone number)
* One voicemail inbox using the enterprise voicemail platform
* Abbreviated (e.g., four-digit) intra-company dialing from the mobile phone
* Class-of-service extended to mobile calls for better control
* Mobile call tracking/logging by the telecom manager using the PBX
Savings
* Outbound mobile calls routed through the enterprise PBX are “on-net” and included in the monthly service fee. (Inbound calls to the mobile phone do incur minutes.)
* Mobile-to-international calls are billed as if from the enterprise PBX or VPN
* Desk phones can be eliminated if desired
* Billed as an add-on feature ($20/month) to an existing Sprint CDMA Wireless Plan
Requirements
* Premise-based Avaya Communications Manager (IP or TDM)
* Sprint CDMA mobile phone with data capability
* Sprint Dedicated IP or Global MPLS VPN connection
More details on Sprint Wireless Integration are available at Sprint.com/voip.
Links;
Businesswire your daily news source
Sprint wireless integration
Source: snapvoip.blogspot.com
According to a news release by Pika technologies, Intuitive Voice, a customer of PIKA, has introduced a low-cost, high-quality PBX phone system designed for small businesses. The Evolution PBX 2.3 combines the cost benefits of open-source PBX technology with the quality of PIKA voice processing, along with an easy-to-use interface designed especially for small businesses with no technical staff.
Chris Jones, President of Intuitive Voice, said, “With Evolution PBX, there are no barriers to prevent small businesses from realizing the benefits of big-business PBX features. At less than $1,500US for a four-line PBX, Evolution PBX is truly affordable for a small business, without sacrificing voice quality or usability.”
For a fraction of the price of traditional PBXs, Evolution PBX provides small businesses with features such as an automated attendant; unique extensions and voicemail boxes for each employee; automatic call routing; integration with Microsoft® Outlook™ and voicemail-to-email; on-hold music; support for telecommuters, voice-over-IP, and much more.
Ease-of-use is assured with the Evolution PBX turnkey software interface, which automates setup and provisioning, and allows non-technical staff to easily maintain the system.
Adds Jones, “These breakthroughs were made possible by PIKA’s high-quality voice processing for Asterisk. PIKA’s low density analog board is unique because it provides advanced functionality such as DSP-based echo cancellation, fax, and native switching, at a very competitive price point.”
Terry Atwood, Vice President of Sales, Marketing and Customer Care at PIKA Technologies, added, “We are very pleased that Intuitive Voice has leveraged our landmark PIKA Connect for Asterisk software to fill a major gap in the PBX market. While the open-source Asterisk solution makes PBX technology affordable, only a company like Intuitive Voice can develop it into a fully-featured and easy to use product that makes sense for small business end users.”
Source: snapvoip.blogspot.com
IBM has developed a complete VoIP service package together with it’s telephony business partner, Vocaltec, based entirely on open standards, Please note, not Open Source as many systems I attribute here. But it is a part of our domain and some customers demand assurance that company like IBM could afford to offer them. The main artery, I think it is the heart, of this solution is the application server Essentra BAX™, VocalTec. Essentra BAX™ features a three-tier architecture, consisting of a database, Web and Call Control & Feature servers. It runs on an IBM eServer x336 or IBM Blade Center under Red hat Linux. The x336 system already supports up to 20,000 users. But that does not stop there, the solution is scalable up to 2 million users, as customers are able to distribute the various components of the three-tier system on a number of X-Series servers or Blades, like database on one or two servers, Web control on another or a server farm, Call control on one or more servers etc. Even larger server farms like this could be managed as easy as a single server, thanks to the design of the system. Once deployed, communications take place within customer premises over own LANs or WANs, via SIP. To connect to existing PBX’s or PSTN is conducted through Voice Gateways. A Session Border Controller (SBC) is provided for integrating with existing VOIP networks or other VOIP providers. If the customer requires more than the provided services, there are interfaces and APIs for services such as application servers, Voice mail, conferencing or Billing features. The IBM / Essentra-Bax solution offers a broader array of features and services, such as Boss Secretary Functions, Hunt Group or Attendant Console, required to manage telephony systems within an organization. The icing on the cake is, services by IBM, which offers one stop service for hardware and software procurement and implementation. This includes roll out Services, operations , maintenance or project management.
Source: snapvoip.blogspot.com
IBM has developed a complete VoIP service package together with it’s telephony business partner, Vocaltec, based entirely on open standards, Please note, not Open Source as many systems I attribute here. But it is a part of our domain and some customers demand assurance that company like IBM could afford to offer them. The main artery, I think it is the heart, of this solution is the application server Essentra BAX™, VocalTec. Essentra BAX™ features a three-tier architecture, consisting of a database, Web and Call Control & Feature servers. It runs on an IBM eServer x336 or IBM Blade Center under Red hat Linux. The x336 system already supports up to 20,000 users. But that does not stop there, the solution is scalable up to 2 million users, as customers are able to distribute the various components of the three-tier system on a number of X-Series servers or Blades, like database on one or two servers, Web control on another or a server farm, Call control on one or more servers etc. Even larger server farms like this could be managed as easy as a single server, thanks to the design of the system. Once deployed, communications take place within customer premises over own LANs or WANs, via SIP. To connect to existing PBX’s or PSTN is conducted through Voice Gateways. A Session Border Controller (SBC) is provided for integrating with existing VOIP networks or other VOIP providers. If the customer requires more than the provided services, there are interfaces and APIs for services such as application servers, Voice mail, conferencing or Billing features. The IBM / Essentra-Bax solution offers a broader array of features and services, such as Boss Secretary Functions, Hunt Group or Attendant Console, required to manage telephony systems within an organization. The icing on the cake is, services by IBM, which offers one stop service for hardware and software procurement and implementation. This includes roll out Services, operations , maintenance or project management.
Source: snapvoip.blogspot.com
IBM has developed a complete VoIP service package together with it’s telephony business partner, Vocaltec, based entirely on open standards, Please note, not Open Source as many systems I attribute here. But it is a part of our domain and some customers demand assurance that company like IBM could afford to offer them. The main artery, I think it is the heart, of this solution is the application server Essentra BAX™, VocalTec. Essentra BAX™ features a three-tier architecture, consisting of a database, Web and Call Control & Feature servers. It runs on an IBM eServer x336 or IBM Blade Center under Red hat Linux. The x336 system already supports up to 20,000 users. But that does not stop there, the solution is scalable up to 2 million users, as customers are able to distribute the various components of the three-tier system on a number of X-Series servers or Blades, like database on one or two servers, Web control on another or a server farm, Call control on one or more servers etc. Even larger server farms like this could be managed as easy as a single server, thanks to the design of the system. Once deployed, communications take place within customer premises over own LANs or WANs, via SIP. To connect to existing PBX’s or PSTN is conducted through Voice Gateways. A Session Border Controller (SBC) is provided for integrating with existing VOIP networks or other VOIP providers. If the customer requires more than the provided services, there are interfaces and APIs for services such as application servers, Voice mail, conferencing or Billing features. The IBM / Essentra-Bax solution offers a broader array of features and services, such as Boss Secretary Functions, Hunt Group or Attendant Console, required to manage telephony systems within an organization. The icing on the cake is, services by IBM, which offers one stop service for hardware and software procurement and implementation. This includes roll out Services, operations , maintenance or project management.
