Wednesday, May 31, 2006

Snom and Digium

Digium, the developer of the Open Source telephony system Asterisk, has worked with German IP Phone manufacturer snom to ensure interoperability of the desk phones with Digium’s Asterisk Business Edition.

The snom 360 and snom 320 phones, available in Australia from Alloy partners, are a SIP-based family of IP telephones engineered and priced for the SME market. In announcing as a Digium Premier Interoperability Partner, snom can assure the phones provide access to the rich feature sets of the Asterisk IP PBX.

“By partnering with innovative companies like snom, we can offer the business community more options and features when building high-end telephony systems based on Asterisk,” said Jim Webster, director of Software Technologies at Digium.

“We look forward to working together with snom to expand the VoIP market and at the same time communicate the important role open source will play in the telecom market.”

The Asterisk Business Edition, the professional-grade version of Asterisk runs on industry standard servers equipped with PCI telephony interface cards. This highly cost-effective approach combines Voice over IP, TDM, switched and Ethernet architectures in one solution.

Digium provides quality hardware and software products that enable telephony applications including legacy PBX, IVR, auto attendant, next generation gateways, media servers and application servers. Digium also offers a full range of professional services including consulting, technical support and custom software development services.

The code for Asterisk was originally written by Mark Spencer of Digium and was continued to the open source world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323.

snom
Digium

Thursday, May 25, 2006

Astricon Tour Europe 2006

Astricon returns to Europe this summer with a three-city tour. Events will be held in London, Paris and Berlin. The program at each tour stop will include top-notch technical speakers, a developer meeting, an exhibition of Asterisk products and services, an introductory seminar, and plenty of local content.

Click for more information about Astricon Europe Tour.

Click to sign up for Astricon Europe Tour.

Introducing the UTstarcom F3000 Wifi Clamshell

This incredibly stylish SIP- WiFi phone has been released-

The new residential Wi-Fi handset is a clamshell type design that supports 802.11b/g, SIP, SDP, RTP, DHCP and TFTP.

http://www.digiumcards.com/utstarcom_f3000.html

Make us an offer by clicking live support


Asterisk

Wednesday, May 24, 2006

Gizmo Project 2.0 Works with Asterisk PBX VoIP

VoIP Soft Phone can be downloaded free @
http://gizmoproject.com/asterisk.html

Sunday, May 21, 2006

Linksys Ships Wireless G VoIP Phones

IRVINE, Calif., May 17 /PRNewswire/ -- Linksys(R), a Division of Cisco Systems, Inc., the recognized leading provider of voice, wireless and networking hardware for the consumer, Small Office/Home Office (SOHO) and small business customer, today announced the immediate availability of its WIP300 and WIP330 Wireless-G IP Phones. The WIP300 and WIP330 are the first in a line of Wireless IP telephony products from Linksys that will enable users to make low-cost Voice over IP (VoIP) calls through 802.11g wireless networks.

"Voice over IP has clearly emerged as the future of telephone communication and Linksys is leading the way," said Tarun Loomba, director, product management, Linksys. "With the launch of our family of WiFi phones, we can take the technology to a whole new level."

WiFi phones from Linksys enable high-quality VoIP service through a Wireless-G network and high-speed Internet connection. While both phones provide clear and reliable voice communication over broadband networks, users can select the appropriate phone and Voice over IP Service Provider for their specific needs.

WIP330

An intuitive user interface on the WIP330 allows users to quickly and easily configure the handset to access available Wireless-G networks. With its built-in Internet browser, the WIP330 can attach to wireless networks including public hotspots which require user name or password access. The browser can also be used to access web-based email, view web sites or even receive Internet-based video, such as that sent by the Linksys WVC54GC Wireless-G Compact Internet Video Camera.

Features of the WIP330 include: * Support for SIP v2 standards * Compliance with IEEE 802.11b/g standards * 2.2-inch color LCD display * Support for Quality of Service (QoS) * Enhanced power savings * 3-way conferencing, call hold and resume, and caller ID * Fast Hotspot Authentication * Support for auto-provisioning using HTTP or HTTPS for configuration and upgrades WIP300

SecureEasySetup (SES) functionality on the WIP300 makes establishing a connection with a Linksys Wireless-G router as easy as pressing two buttons, one on the phone and one on the router. A highly secure WPA-encrypted connection is established that allows the user to make VoIP calls from within that network.

Features of the WIP300 include: * Support for SIP v2 standards * Compliance with IEEE 802.11b/g standards * 1.8-inch color LCD display * Call forwarding, call transfer, call history * Backlit keypad * RF and battery level indication * WiFi survey tool * Connection status button * USB charger interface WiFi Phone Requirements

Both the WIP330 and WIP300 require a broadband Internet connection, a wireless router or access point with DHCP server, and activated VoIP service.

Pricing and Availability

The WIP300 and WIP330 are available immediately through authorized Linksys distributors with estimated street prices of $219.99 and $369.99 respectively. Additional products from the Linksys family of WiFi phones will be launched in the second half of 2006.

About Linksys

Founded in 1988, Linksys, a Division of Cisco Systems, Inc. www.linksys.com is the recognized leader in Voice, Wireless and Ethernet networking hardware for consumer, SOHO and small business users. Linksys is dedicated to making networking easy and affordable for its customers, offering innovative, award-winning products that seamlessly integrate with a variety of devices and applications. Linksys provides award-winning product support to its customers. For more information, visit .

Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and products are trademarks or registered trademarks of their respective holders. Copyright (C) 2006 Cisco Systems, Inc. All rights reserved.

CONTACT: media, Karen Sohl, +1-949-823-1578, ksohl@cisco.com, or TrevorBratton, +1-949-823-1212, , both of Linksys; orinvestors, Ken Bond of Cisco, +1-408-526-6001, , for Linksys

Web site: http://www.linksys.com/

Thursday, May 18, 2006

Linksys SPA3102

The SPA3102 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via the advanced implementation of standard voice coding algorithms. The SPA3102 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems.The SPA3102 offers all the key features and capabilities which service providers can provide customized VoIP services to their subscribers. The SPA3102 can be remotely provisioned and supports dynamic, in service software upgrades. A secure profile upload saves providers the time, expense, and hassle of managing and pre-configuring or reconfiguring customer premise equipment (CPE) for deployment.Linksys understands that security for end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The SPA3102 supports secure, standard encryption-based methods for communication, provisioning and servicing.

Wednesday, May 17, 2006

WRTP54G By Linksys VoIP



All-in-one Internet-sharing Router, 4-port Switch, and 54Mbps Wireless-G (802.11g) Access Point
Shares a single Internet connection and other resources with Ethernet wired and Wireless-G and B devices
Two standard phone jacks enable feature-rich telephone service over your cable or DSL Internet connection
High security: Wi-Fi Protected Access™ (WPA), wireless MAC address filtering, powerful SPI firewall

The Linksys Wireless-G Broadband Router is really four devices in one box. First, there's the Wireless Access Point, which lets you connect both screaming fast Wireless-G (802.11g at 54Mbps) and Wireless-B (802.11b at 11Mbps) devices to the network. There's also a built-in 4-port full-duplex 10/100 Switch to connect your wired-Ethernet devices together. Connect four PCs directly, or attach more hubs and switches to create as big a network as you need. The Router function lets your whole network share a high-speed cable or DSL Internet connection.

The fourth function is the phone adapter which enables high-quality feature-rich telephone service through your high-speed connection even while you're surfing the Internet. There are two standard telephone jacks, each operating independently -- like having two phone lines. With Vonage Service, you'll get low domestic and international phone rates, Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and lots of other available special phone features. Choose any free local dialing US area code, regardless of where you live. Add a virtual phone number in any area code, or even a US-wide toll-free number.

To protect your data and privacy, the Wireless-G Broadband Router can encode all wireless transmissions with up to 128-bit encryption, and supports both Wired Equivalent Privacy (WEP) and the industrial-strength wireless security of Wi-Fi Protected Access™ (WPA). The Router can serve as a DHCP Server, has a powerful SPI firewall to protect your PCs against intruders and most known Internet attacks, supports VPN pass-through, and can be configured to filter internal users' access to the Internet. Configuration is a snap with the web browser-based configuration utility.

With the Linksys Wireless-G Broadband Router at the center of your home or office network, you can share a high-speed Internet connection, files, printers, and multi-player games, and turn that Internet connection into a high-quality, high-value telephone service!

Linksys WRTP54G - Wireless VoIP



-All-in-one Internet-sharing Router, 4-port Switch, and 54Mbps Wireless-G (802.11g) Access Point
-Shares a single Internet connection and other resources with Ethernet wired and Wireless-G and B devices
-Two standard phone jacks enable feature-rich telephone service over your cable or DSL Internet connection
-High security: Wi-Fi Protected Access™ (WPA), wireless MAC address filtering, powerful SPI firewall

The Linksys Wireless-G Broadband Router is really four devices in one box. First, there's the Wireless Access Point, which lets you connect both screaming fast Wireless-G (802.11g at 54Mbps) and Wireless-B (802.11b at 11Mbps) devices to the network. There's also a built-in 4-port full-duplex 10/100 Switch to connect your wired-Ethernet devices together. Connect four PCs directly, or attach more hubs and switches to create as big a network as you need. The Router function lets your whole network share a high-speed cable or DSL Internet connection.

The fourth function is the phone adapter which enables high-quality feature-rich telephone service through your high-speed connection even while you're surfing the Internet. There are two standard telephone jacks, each operating independently -- like having two phone lines. With Vonage Service, you'll get low domestic and international phone rates, Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and lots of other available special phone features. Choose any free local dialing US area code, regardless of where you live. Add a virtual phone number in any area code, or even a US-wide toll-free number.

To protect your data and privacy, the Wireless-G Broadband Router can encode all wireless transmissions with up to 128-bit encryption, and supports both Wired Equivalent Privacy (WEP) and the industrial-strength wireless security of Wi-Fi Protected Access™ (WPA). The Router can serve as a DHCP Server, has a powerful SPI firewall to protect your PCs against intruders and most known Internet attacks, supports VPN pass-through, and can be configured to filter internal users' access to the Internet. Configuration is a snap with the web browser-based configuration utility.

With the Linksys Wireless-G Broadband Router at the center of your home or office network, you can share a high-speed Internet connection, files, printers, and multi-player games, and turn that Internet connection into a high-quality, high-value telephone service!

Wireless VoIP

Cisco Systems' Linksys networking gear unit is venturing into new territory. The company unveiled its first phones designed to enable customers to leverage 802.11g wireless networks to make voice over IP (VoIP) calls.

The new product line includes two Wireless-G IP phones: the WIP300 and WIP330. The WIP330 allows users to configure the handset to access available Wireless-G networks. The WIP330 comes equipped with a built-in Internet browser that enables it to attach to wireless networks, such as public hot spots that require user name or password access.

The WIP330's SecureEasySetup makes connecting with a Linksys router a 2-button job. Once secure WPA-encrypted connection is established, the WIP330 user can make VoIP calls from within that network.

In January, Linksys teamed with Yahoo! to introduce the Linksys Wireless-G Music Bridge, which was designed to complement Yahoo's digital music download service.

Last year, Linksys began selling its Wireless-G product line at OfficeMax retail outlets. The agreement was part of Linksys' goal of focusing on solution-based merchandising and small business-related market initiatives, according to the company.

The WIP300 and WIP330 are priced at $219.99 and $369.99, respectively. Linksys plans to expand its line of Wi-Fi phones in the second half of 2006.

IP Telephone Equals Business Productivity

IP telephony (internet protocol-based phones) had a bumpy beginning with problems of voice quality plaguing its existence in the first five to six years. But about two to three years ago, networking and telecommunication companies started re-focusing on IP telephony and it turns out that IP phones are now making serious inroads into business telephony the world over. Prabhakar Deshpande caught up with Lawrence Byrd, Director, IP Telephony and Mobility, Avaya, and talks about the growing trend.

How is IP telephony evolving?

More than 50% of our global shipments of telecommunication equipment for 2005 have been IP-based. In India, we expect this shift to likely happen in 2006. Admittedly, the installed base of business telephony - over 450 million lines -- is still likely to be based on traditional telephony.

Why should one shift to IP telephony?
We allow customers to evolve at their own pace. This is not an all-or-nothing approach. The reason IP telephony is being advocated is that it allows for new applications and contributes to productivity. There are new applications in the area of mobility, conferencing and unified access that would be very difficult to implement using traditional telephony at reasonable costs.

How would this shift to IP telephony impact vendors?
The movement to IP-based phones would certainly impact vendors. Avaya had to invest significantly for a major technology shift. So did other vendors. Carriers such as AT&T, hitherto working in traditional telecom, cannot be complacent and would have to provide alternative service offerings. Many smaller PBX companies are in fact, disappearing.

Why did this movement to IP happen?
The movement to IP telephony started happening in the late 90s. Companies found that they had data fibres for data networks and leased lines for traditional voice network. The vendor push in this area was that there was no need to separate the two whereby costs could also actually be reduced. There was a similar customer push as IP Telephony was cheaper and allowed richer set of applications.

What are the rich sets of applications?
Many applications are in the area of video conferencing, web conferencing and unified messaging. One advantage is in the area of providing access to many more people while another advantage is being able to use a cellphone as an extension to IP. Video over IP is yet another application and it is also possible to have an office phone at home with a virtual private network.

Plantronics Introduces VoIP-Optimized Voyager 510-USB

Plantronics launched the Voyager 510-USB, a Bluetooth headset system optimized for Voice over Internet Protocol, VoIP, providing instant wireless connectivity to PC-based softphones. The company noted that the Voyager 510-USB reflected its commitment to developing innovative VoIP products for both business and consumer applications. The company noted that the system included a plug-and-play Bluetooth USB adapter that eliminated cumbersome software setup processes and enabled mobile professionals to place hands-free calls through any VoIP service. In addition, Voyager 510-USB featured multipoint technology to allow users to switch between multiple Bluetooth-enabled devices, including softphones and mobile phones.
Please Visit
http://www.digiumcards.com for Plantronics VoIP gear!

Tuesday, May 16, 2006

PIKA Connect For Asterisk Now Supports Asterisk 1.2 and Low-Density Boards


Media processing hardware and software manufacturer PIKA Technologies announced today the latest release of PIKA Connect for Asterisk, which adds support for PIKA’s low-density analog boards and compatibility with Asterisk 1.2.

PIKA Connect is a free software layer/channel driver, distributed under the GNU Public
License, that provides connectivity between PIKA’s boards and Asterisk’s open-source PBX software.

In a press release, PIKA architect Wojciech Tryc commented that, “With PIKA Connect for Asterisk, the Asterisk development community can benefit from advanced features for fax and echo cancellation in low and high density analog applications.”

Tryc continued: “In addition, support for H.100 allows for low latency native switching between our high density analog boards. This is made possible by PIKA's DSP processing power on the board.”

The new version of PIKA Connect for Asterisk will be available by the end of June, 2006, on the PIKA Developer Zone, the company said.

Friday, May 12, 2006

The VoIP Key System

New all-in-one VoIP and analog phone/communications system from Allworx® gives small businesses, for the first time ever, a fully featured VoIP key system replacement option for their traditional (TDM) key system – at a comparable price point.

Allworx® 6x is a three in one platform: a rich phone system, a robust data network and a message center. It offers a host of features and benefits that make it ideal for offices up to 30 employees, including all the traditional key phone system features critical to SMB users.

It also includes a built-in, eight-way conference-bridge system and options for Allworx Call Queuing, VPN, calendaring, expanded voicemail and the new Allworx Call Assistant software.

Allworx® phones are designed as plug-and-play solutions that automatically “phone home” to the 6x system allowing remote users to seamlessly take advantage of all the system features while out of the office.

http://www.digiumcards.com
231.946.4162

AudioCodes Announces a Residential/SOHO Gateway for VoIP Telephony Service MP-202

The MP-202 Telephone Adapter, a member of the MediaPack™ gateway series, is based on Audiocodes' perfecto core tecnology, benefiting from advanced voice technologies for an enhanced voice quality and user experience.

The MP-202 is a 2 line SIP gateway allowing users to connect to ordinary POTS telephone or fax machines, and it is interoperable with leading softswitches and SIP Application servers for enbiling legacy phone services such as caller ID, call waiting and call fowarding. In addition the MP-202 includes an internal router with DHCP, NAT and PPoE capabilities enabiling suscribers to connect their home PC or LAN hub/switch to it.

Audiocodes is expecting to start shipping these units by the end of June 2006

Introducing the IX68 with ADSL2+modem

The SurfinBird IX68 is a new generation of SIP aware combo IAD router and firewall all in one. It takes the previous rewarded Intertex ADSL Modems to new dimensions. The new SurfinBird IX68 models have higher capacity, better functionality and more IP telephony features based on the SIP standard. This device incorporates NAT traversal or ALG properties with a solid QoS and is a ideal unit for VoIP Service Providers playing in the ADSL space. Works greate with snom IP Phones.

It comes with the following options: wireless access point supporting IEEE 802.11b/g 54 Mbps, telephony ports for connecting a conventional telephone for use with SIP based IP Telephony or to work as a gateway for SIP based IP telephony and the old telephone network and SIP switch (for PBX like features).

A version that supports a single FXO port and FXS port are available. Ideal for hosted PBX players that need a redundancy switch and need a immediate solution to routing 911 and local calls.

IX68 -Complete Broadband CPE/IAD!

Complete Provisioning Service Available for Linksys Phones

Linksys adds new IP Phones and IP devices. The new 942 now offers POE which makes it a phone suitable for office deployment Linksys 942 IP Phone. This is also a great Service Provider product, for IP Centrex / Hosted PBX applications. These solid looking SOHO style IP Phone with 2 lines that can be field upgraded to 4 lines, a second ethernet port and power over ethernet. (POE) IEEE 802.3af. Stylish design with 4 LED lit buttons for VM, hold and other functions.

ABP also introduced the SPA901 and expecting to have soon the new phones SPA921, SPA922 and the Linksys wireless phone WIP300.

The SPA901, small, affordable, single line business class IP Phone, wall mount or table top phone. It comes with easy installation and secure remote provisioning. Menu based and web based configuration.

Ideal products for ITSPs that are already deploying Linksys product as ATAs and whish to have a consistent provisioning system across the platform.

Epygi Quadro Has New Features 2X, 4X, and the 16X

Using the new 3.1 software, a Quadro can be configured to connect all calls to the receptionist.

If the receptionist is on the phone at the time a call comes in, the second call will be answered by a queue and a hold message will be played, "Please hold, someone will be available to answer your call momentarily," then hold music will play. Once the receptionist is off the first call, his/her phone will ring, and when answered, the first caller in the queue in FIFO order will be connected to the receptionist. Should the receptionist not answer the call when it comes out of the queue, it will go to voicemail for the receptionist's extension.

A virtual extension can be created that contains a group of extensions in a 'pickup group'. Any phone in the office can pick up any calls to these extensions by dialing the virtual extension from these other phones.

Paging and full support for snom IP phones including line appearances are now included.

Guaranteed Internet Access for your VoIP System

Can you run your business with out significant pain and distraction when your Internet link is down and you have no email and no VoIP?

Give your customer peace of mind that his business operations will never be without internet access by using our new Astrocom PowerLink Pro™ to create a redundant Internet access path by combining 3 to 15 Internet connections aggregating bandwidth and load balancing.

In some regions you just can't get a data T1 or its simply too expensive. With this product you can bind several business DSLs and/or cable modem links to exceed the performace of a T1 with better redundnacy.

PowerLink Pro™ is a multi-homing Internet access appliance, an Internet traffic manager if you will, that sits between the LAN and the WAN. Combine Cable, T1/E1s, ADSLs or and even multiple ISDN lines for a more powerful and redundant Internet link. This is great for VoIP but can also be a life saver for users of hosted applications like CRM or ERP systems.
Professional deployments have a plan for redundancy. Once your customers understand this product they will never give up the peace of mind this offers them

Cisco Announces the Appointment of Michael Pocock

Pocock Brings Extensive Consumer Technology Experience; Victor Tsao and Janie Tsao to Focus Efforts on New Opportunities in China for Cisco

/noticias.info/ IRVINE, CA - May 11, 2006 - Linksys®, a division of Cisco Systems, Inc., the leading provider of VoIP, wireless and networking hardware for the consumer, SOHO (Small Office/Home Office) and small business environments, today announced that J. Michael Pocock has been appointed to the position of Linksys senior vice president and general manager. Victor Tsao and Janie Tsao, Linksys founders who have shared this SVP/GM role since 2003, will now focus their efforts on new opportunities in China for Cisco.

In his new role, Pocock will oversee worldwide Linksys operations, sales, marketing, human resources, training, customer advocacy and all product development and engineering. Pocock will focus his efforts on leading Linksys' strategy in the consumer and small business markets, while continuing to expand into new markets and grow the business on a world wide basis.

Pocock most recently served as the President and Chief Executive Officer of the Polaroid Corporation, a world leader in instant camera and film products. Pocock joined Polaroid in 2003 and during his tenure, Polaroid become the world's top seller of portable DVD players and entered MP3 handheld and plasma TV markets. Prior to taking the position at Polaroid, Pocock served in leadership roles at Xerox Corporation, Epson, Digital Equipment Corporation, Compaq and General Electric. Pocock will report into Charles Giancarlo, Cisco SVP, chief development officer and Linksys president.

"This is an ideal time to have Michael Pocock join the Linksys Division of Cisco Systems. In an era when technology is revolutionizing the way we enjoy and interact with music, television, and games, and communicate, manage and share personal information, his experience and leadership in the consumer space will help take Cisco to the next level in the global home networking market," said Charles H. Giancarlo, chief development officer and president of Linksys. "We were privileged to have the founders of Linksys, Janie and Victor Tsao, on our leadership team as we made great strides in the innovation of home networking over the past few years. Now we have the opportunity to leverage their entrepreneurial talent in a new business capacity as they transition to business development roles in China."

"It is an exciting opportunity to be able to lead the Linksys team," said Michael Pocock, Linksys SVP and GM. "Linksys is widely respected in the consumer channel with its strong reputation for being an easy company to do business with and for bringing high-quality and innovative products quickly to market. This is the kind of company that can continue to make a difference in the way people communicate and access information and I am honored to carry on the tradition that Victor and Janie have built over the past 18 years."

Victor and Janie Tsao will continue to report into Charles Giancarlo as they develop new business opportunities for Cisco in China. The Tsaos will seek out business investment opportunities, work to collaborate and build relationships with local businesses, and explore new product markets for Cisco and Linksys.

"We felt the organization was ready for a new leader who would take Linksys to its next level," said Janie Tsao, Linksys co-founder and SVP. "We have more than doubled our revenue since we were acquired in 2003 and we think the organization will benefit from someone with more experience in managing multi-billon dollar corporations. After 18 years of growing with the company, it's bittersweet to hand it off, but Victor and I are looking forward to sharing our experience with Cisco's customers and colleagues in China and bringing additional value to the entire Cisco organization."

Michael Pocock formally started his new position in Irvine, California on May 8, 2006. The Tsaos will stay at Linksys until Pocock's transition is complete and will then serve as consultants to the Linksys organization.

About Linksys
Founded in 1988, Linksys, a division of Cisco Systems, Inc., (NASDAQ: CSCO) is the recognized global leader in Wireless, VoIP and Ethernet networking for consumer, SOHO and small business users. Linksys is dedicated to making networking easy and affordable for its customers, offering innovative, award-winning products that seamlessly integrate with a variety of devices and applications. Linksys provides award-winning product support to its customers. For more information, visit

www.linksys.com.
VoIP - Telephony

Thursday, May 11, 2006

Ranch Networks Develops Redundancy Solution for Asterisk

New Feature Enables Open Source PBX Users Uninterrupted VoIP Service


Morganville, NJ – May 8, 2006 – Ranch Networks, the first IP telephony network appliance provider to integrate security and bandwidth control for IP-based applications, today introduced 1+1 High Availability (HA) to its RN series of appliances. The 1+1 HA feature will provide users with reliable, redundant and uninterrupted VoIP service between any two Asterisk servers, even when the servers are not on the same network.

When connecting two Asterisk servers to a RN appliance, the RN appliance monitors both servers through the 1+1 HA solution by simulating real-time SIP registration requests. If one of the servers stops responding to the SIP requests, the RN appliance will automatically redirect all calls to the other Asterisk server—providing users with uninterrupted VoIP service.


The 1+1 HA feature enables synchronization among the two Asterisk servers by representing them as a single IP address, which allows IT personnel to maintain or replace any of the Asterisk servers without VoIP service interruption. Ranch Networks developed the 1+1 HA solution to synchronize the databases in both Asterisk IP PBX servers allowing branch office users continued and uninterrupted VoIP service.

“Ranch Networks understands the need for reliable VoIP service,” said Ram Ayyakad, founder and CEO of Ranch Networks. “The 1+1 HA feature enables the Asterisk IP PBX servers to provide reliable 24/7 VoIP service to enterprise level solutions.”
Starting with Asterisk, the open source PBX, Ranch Networks’ offers the RN300, RN20, RN40 and RN41 to provide dynamic, protocol independent, per-call authenticated network access. These products will supply unprecedented VoIP security, bandwidth management, VPN, accounting and switching capabilities to small to mid-size enterprises, service providers and carriers.


The RN300, RN20, RN40, and RN41 are available through leading Asterisk resellers worldwide ranging in price from $600 to $17,300. For a list of resellers or additional information, please visit www.ranchnetworks.com.


About Ranch Networks
Ranch Networks offers IP telephony network appliances that secure, manage and scale VoIP traffic beyond existing firewall technologies. The company’s flagship RN series of products are the first-ever appliances to integrate security and bandwidth control for VoIP applications. Established in 2000 by award-winning ex-Bell Labs engineers, Ranch Networks guarantees the delivery of reliable and secure IP communications between office employees, remote office locations or service providers and enterprise customers.




About Asterisk
Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports US and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.



Ranch Networks and the Ranch Networks logo, are registered trademarks of Ranch Networks, Inc. All other trademarks are the property of their respective owners.

Indosoft Deploys Asterisk-based Hosted Multi-Tenant Contact Center for VirtuServe

Indosoft has recently completed the installation of its Asterisk based multi-tenant Contact Center for VirtuServe, a Chicago based nationwide provider of hosted, multi-tenant services for Enterprises.

Fredericton, NB (PRWEB) May 11, 2006 -- Rapid advancement in Voice over IP technology is creating a paradigm shift in the way Contact Centers handle voice and data traffic. Businesses are looking towards Asterisk PBX and multi-tenant Contact Center solutions to lower infrastructure and setup costs. This insulates Organizations from the complexities and costs of operating and managing a contact center.

Indosoft has recently completed the installation of its Asterisk based multi-tenant Contact Center for VirtuServe, a provider of hosted, multi-tenant services for Enterprises. Indosoft’s Contact Center solution is tightly integrated with Asterisk IP-PBX. Along with web-based CRM to manage the Agent and Administration interface, Indosoft provides all the essential components for an Asterisk PBX based fully blended call center solution.

• Indosoft Q-Pump is a sophisticated session manager capable of providing blended solution.
• Indosoft Multi-Connector is proxy to the Asterisk Manager Interface, which enables CTI-like functions for all non-telephony software sub-components.
• The Indosoft Inbound GUI tool will setup call management and routing for your Inbound ACD requirements by leveraging the complete potential of Enterprise-grade Asterisk Dial-plan and Asterisk Queues.
• Indosoft Predictive Dialer technology is designed to deliver greater contact rate and ‘talk to’ time while working within FCC guidelines.
VirtuServe (http://virtuserve.com) provides quality customer service agents for outbound telemarketing, and inbound customer support, for $12/hour by leveraging open source Asterisk and Indosoft Contact Center Technology.

Indosoft Inc. (http://www.indosoft.com) offers complete call centre solutions on Asterisk IP-PBX with installations all over the world. Indosoft is a Digium Asterisk Partner and provides fully blended solutions for Contact Centers, Audio Conferencing Bridging, Real-time call blocking for Do Not Call list enforcement, as well as Hosted PBX, IVR and Recording solutions. Digium (http://www.digium.com) Quad PRI (E1/T1) boards for Asterisk PBX deliver a cost effective reliable TDM interface to Telco.

Monday, May 08, 2006

VoicePulse Launches API for Asterisk Integrators

Source: VoicePulse Inc

VoicePulse Launches API for Asterisk Integrators
Company Becomes First Asterisk-Friendly Service Provider to Publish API
JAMESBURG, N.J., May 8, 2006 (PRIMEZONE) -- VoicePulse Inc. today announced the immediate availability of an application programming interface (API) for the VoicePulse Connect for Asterisk service. The API allows developers of Asterisk-based products to seamlessly integrate with VoicePulse's provisioning and billing systems.


Asterisk is a complete PBX in software. It runs on Linux, BSD and Mac OSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

"Mature, stable products based on Asterisk are now available," said Ravi Sakaria, VoicePulse's president. "These user-friendly PBXs with intuitive user interfaces have been missing one piece -- tight integration with the service provider. Using the VoicePulse Connect for Asterisk API, developers of these products can, for the first time, develop a truly self-configuring phone system."

VoicePulse Connect for Asterisk customers can make and receive calls from anywhere in the world using the standard SIP protocol as well as Asterisk's own IAX protocol. Using the API, vendors can develop systems to auto-configure the PBX as well as allow customers to add and remove phone numbers, make payments and check their balance from within the PBX's own user interface.

The VoicePulse Connect for Asterisk service includes:


-- Fast, easy to use website at http://connect.voicepulse.com/
-- Incoming U.S. phone numbers from 40 states, 200 area codes,
3,500 cities
-- Incoming toll free phone numbers
-- Asterisk-friendly website, configuration samples and
customer support
-- Prepay (pay-as-you-go) service
-- No volume commitment or activation fees to open account
-- 2.4 cents/min outgoing to the U.S. 48
-- 3.9 cents/min incoming toll free
-- $11/month for each incoming phone number
-- Free incoming minutes

The VoicePulse Inc. company logo is available at http://www.primezone.com/newsroom/prs/?pkgid=1724

CONTACT: VoicePulse Inc.
Chris Liu
732-339-5100
chris.liu@voicepulse.com

Saturday, May 06, 2006

SIPphone.com Joins DIDxChange.com

DIDx provides "Virtual Numbers" for Gizmo Project European VoIP customers; users of the popular Internet calling software benefit from local phone numbers.

Pensacola, FL, May 06, 2006 --(PR.COM)-- Super Technologies today announced that SIPphone, the VoIP platform and directory behind the popular Gizmo Project Internet calling software, is using its DIDx service to provide local phone numbers for several countries in the European Union.

Through Super Technologies' DIDxchange global DID ("direct inward dialing") marketplace, Gizmo Project now offers "Virtual Numbers" for customers in France, UK and Spain. These numbers enable users to have their own traditional, local phone number for VoIP-enabled devices and "softphones" like Gizmo Project. Using an easily recognizable local phone number, friends and business contacts can reach Gizmo users no matter where they happen to be physically located. Additional "Virtual Numbers" for international customers will be rolled out in the future.

"The Internet opens geographic boundaries so that the idea of doing business inside just one country is being superseded by a focus on competing in the global marketplace," said Super Technologies, Inc. DIDx VP Suzanne Bowen. "DIDx assists the best Internet Telephony Service Providers such as Gizmo Project to do exactly that. Gizmo will now be able to automatically sell the DID's from DIDxchange under the average consumer target price. Global carriers can now make cutting edge companies such as SIPphone their customer simply by joining the DIDxchange."

The Gizmo Project uses an Internet connection to allow users to make free phone calls using a Windows, Macintosh or Linux personal computer. Gizmo also sports features like conference calling, voice-mail and instant messaging (IM) that is completely inter-operable with Google Talk and hundreds of other IM networks. Inexpensive add-ons like Call In and Call Out allows Gizmo callers to connect to others who may be using landlines and cell phones.

"The number one feature request for Gizmo we're hearing from International users is to have a familiar, local number that their friends, family and work associates can dial," said Jason Droege, President of SIPphone. "Tapping into the DIDx service for these and future `virtual' numbers will fuel growth of the Gizmo Project worldwide."

The Gizmo Project is freely available from http://www.gizmoproject.com.

About Super Technologies DIDxchange:

Super Technologies, Inc. was founded in 1999 with the objective to give leverage to its global customer base of the most practical and innovative voice technology, including white label VoIP software and DIDx. Today, the company connects over 2000 ITSP's, CLEC's and ILEC's to buy, sell and trade DID's (Direct Inward Dialing) with DIDxchange and others. For further information, visit http://www.didx.org.

About SIPphone:

SIPphone is a San Diego, CA-based Voice over IP (VoIP) company founded in 2003 that provides a VoIP platform and directory ("dialtone") on which any SIP-based hardware or software developer can offer free VoIP service. SIPphone is also a founding member of IM Federation (IMfederation.com), which promotes interconnects between all networks. SIPphone currently connects with Google's Google Talk and more than a hundred other networks.

The free Gizmo Project software for Apple Macintosh, Microsoft Windows and Linux computers is developed by SIPphone. This VoIP "softphone" enables high-quality free calling worldwide using any Internet-connected computer. Gizmo includes free conference calling, customizable voice-mail, Instant Messaging (IM) and a host of other convenient features. Download the Gizmo Project at http://gizmoproject.com.

SOURCE: Super Technologies, Inc.

Super Technologies, Inc., Pensacola
Suzanne Bowen, 212-736-3719
care@didx.org
or
SIPphone, Inc.
Kevin La Rue, 858-356-5451
Gizmo: KLR
kevin.larue@sipphone.com

Friday, May 05, 2006

VoIP Bandwidth Test

AudioCodes

AudioCodes recently added a new line of analog gateways and .e4 is stocking all units that are available at this time.

The new gateways have some interesting new features and offer a very competitive price point

The current MP104, MP108 and MP124 will still be available although some models are approaching their EOL.

Asterisk: A Bare-Bones VoIP Example

Getting Started With Asterisk

Digium

The TCM400B which is a dedicated codec translation card supporting G.729A and G.723.1. It is a PCI base card with a daughter module, which will support 120-150 simultaneous calls. The next generation card will allow you to put two modules on base card. This is a great card for building gateways or for large enterprise environments.



Digium's New Digital Cards TE420P and TE415P
The TE420P and TE415P cards are loaded with the new echo cancellation card using the Octasic echo cancellation part. These cards provide carrier-grade echo cancellation, Voice Quality Enhancement (VQE), DTMF decoding and tone recognition. They support 128ms of G.168 compliant echo cancellation across all channels.


Digium's B410P 4 Ports BRI Card/ISDN (Latin America)
The new B410P 4 ports BRI card supporting both TE and NT interface. It is a 32-it, 33MHz, PCI 2.2 compliant card and works on 3.3V and 5V PCI slots. The differentiating feature is the on-board echo cancellation.

Astrocom PowerLink

Redundant Internet Link for VoIP with PowerLink Pro
When downtime or slowdowns cost you money protect your Network Performance with PowerLink™ product, a multi-homing Internet access appliance, and an Internet traffic manager, that sits between the LAN and the WAN.

Eicon Networks Diva Server Cards

Robust, intelligent communication hardware adapters for analog, ISDN BRI and PRI, and E1/T1 connections.


The support of open standards (Asterisk) and a single programming API ensure that Diva Server can be used in the widest spectrum solutions and that application can handle calls independently of the underlying network whether is TDM or IP.




Remote Access, Fax and Voice technology as well as certified for Microsoft Speechserver and new free SDK 3.5 for Voice Application development.

.e4
http://www.e4strategies.com

Ranch Networks

When combining Asterisk with Ranch Networks series appliances, the solution provides unprecedented security and scalability to the open source telephony industry.



Ranch Networks' Asterisk add-on appliance offers MIDCOM integration which provides dynamic per-call firewall control, bandwidth management, NAT traversal, and RTP traffic bridging – all supporting encrypted signaling streams. The Patent-Pending technology separates voice, video, and data traffic into multiple, secure zones without having to reconfigure IP addresses. This product will allow Asterisk developers to build systems that require very fast switching local nodes. The Ranch Network unit also can be used with Asterisk to do QoS, multi homing and also replace a basic firewall and router for many small installations.

Converged Access Traffic Management

Converged Access Traffic Management:

The Industry's First Application Performance Management Solution to Enable

Toll-Quality VoIP

The CTM brings advanced application performance management to small branch and large data center locations. It is a powerful IP application solution designed to optimize WAN efficiency while delivering precise and granular application performance guarantees.

Check out .e4's website for more info about the Converged Access IP Traffic Management product line. Call now for more details. 231.946.4162