Number of results 25
for General
31/01/2012 - Need some help understanding VOIP
Hello All!
Recently I was approached by a local company to work with them to improve their technical infrastructure.
I have not worked with VOIP systems before. However I have in the past setup Asterisk and was using it successfully as a dialer.
This client is waiting for a VSC VMSx switch. I'm not familiar with it but they are asking for processing/gateway integration and hardware monitoring.
So with that being said here are a few questions:
1. I have searched around for a VMSx API to be add/remove balances from calling cards, however I cannot find one. Can someone point me in the right direction?
2. Hardware monitoring, can that be setup in VMS? For example, to know when all the trunks are being filled to 90% capacity or when there is no more internet, etc.
3. I'm still confused about something. So they have this softswitch sitting in a local data center. They have clients that have gateways setup in their locations. When I worked with Asterisk, we had a switch with all the ip phones plugged into. That switch was connected to asterisk with just one cable to the secondary lan card. If I don't understand correctly the way it should be setup is.
a. Softphone connects to a switch.
b. The switch to the gateway.
4. What about for those that already have an existing PBX or want a complete PBX setup. From what I understand there is some type of module that you can get which simply extends the PBX to make outgoing calls route through the VMS softswitch in the data center. Is it essentially also a gateway module?
Sorry for the novice questions but I really need to know the answer to these questions :)
Thanks in advance!
Recently I was approached by a local company to work with them to improve their technical infrastructure.
I have not worked with VOIP systems before. However I have in the past setup Asterisk and was using it successfully as a dialer.
This client is waiting for a VSC VMSx switch. I'm not familiar with it but they are asking for processing/gateway integration and hardware monitoring.
So with that being said here are a few questions:
1. I have searched around for a VMSx API to be add/remove balances from calling cards, however I cannot find one. Can someone point me in the right direction?
2. Hardware monitoring, can that be setup in VMS? For example, to know when all the trunks are being filled to 90% capacity or when there is no more internet, etc.
3. I'm still confused about something. So they have this softswitch sitting in a local data center. They have clients that have gateways setup in their locations. When I worked with Asterisk, we had a switch with all the ip phones plugged into. That switch was connected to asterisk with just one cable to the secondary lan card. If I don't understand correctly the way it should be setup is.
a. Softphone connects to a switch.
b. The switch to the gateway.
4. What about for those that already have an existing PBX or want a complete PBX setup. From what I understand there is some type of module that you can get which simply extends the PBX to make outgoing calls route through the VMS softswitch in the data center. Is it essentially also a gateway module?
Sorry for the novice questions but I really need to know the answer to these questions :)
Thanks in advance!
17/01/2012 - XConnect Announces Partnership with Germany's DE-CIX
The partnership will create a secure and scalable federation hub and central carrier ENUM-based routing and number-management platform that will enable hundreds of fixed, mobile, ASP and Web 2.0 operators to interconnect securely end to end on an all-IP basis.
Interconnection via the hub will benefit operators significantly. It will reduce the technical complexity and commercial and operational costs of NGN interconnection, allowing for migration from bilateral, legacy-network TDM interconnects to IP. In addition, operators will be able to securely and multilaterally interconnect to multiple networks via a single IP connection to the hub, supporting the interworking and interoperability of IP voice and multimedia services, such as HD voice, video and Unified Communications.
The ENUM registry will, for the first time, offer a central number-management and number-portability platform for German operators, facilitating ENUM-based number discovery and routing that will enable calls to be delivered accurately and directly across networks.
DE-CIX, a wholly owned subsidiary of the largest European Internet industry association, eco, already provides direct and settlement-free Internet interconnection infrastructure services to more than 400 operators.
The partnership agreement announced today combines DE-CIX’s experience in managing high-volume Internet exchange services and market reach with XConnect’s expertise and global leadership in carrier-neutral NGN interconnection services and technology. XConnect currently operates other national federations in Europe, Asia and Africa, and the company is in talks about establishing additional federations.
28/11/2011 - OnSIP Introduces Phone Certification Program to Recognize Exceptional VoIP Products
The partnership will improve the reach of both phone manufacturers and OnSIP through joint promotional efforts. Certified phone manufacturers will be prominently featured on the OnSIP website.
Manufacturers and developers who are interested in having their phones OnSIP-certified can download and complete the application found at http://www.onsip.com/certification-program. Participating in the program is free, and certified phones are pushed higher in the queue for the company’s growing phone review program.
OnSIP’s hosted VoIP service for business works with a wide variety of VoIP phones available on the market and does not limit the end user to any model or brand. Dozens of reviews of phones are already available online. The certification program goes one step further and recognizes phones that perform exceptionally with OnSIP’s service.
27/09/2011 - Voip-Pal Successfully Completes Beta Testing on its PointsPhone.Com Website
Voip-Pal.Com is
pleased to announce that the company has completed its Beta Test phase of its retail
website www.pointsphone.com.
Voip-Pal is taking orders for its low cost VoIP services on its retail website, www.pointsphone.com. Revenue stream has been re-established and new customers are beginning to place orders. Once customers register and sign-up, they have a choice of placing a call using their cell phone or land line or by downloading a PC Dialer or downloading an Open Source App for their smartphone. Voip-Pal is redesigning its new website and the new website will be ready in the upcoming month. Voip-Pal's own branded PointsPhone Mobile Apps for the Android, iPhone, Blackberry and other smartphones are currently being tested on the new North American Cloud Server and will be posted soon on the website.
"The Beta Site notice has been taken off the website and the Free Trial period will end this week," states Dennis Chang, President of Voip-Pal.
07/09/2011 - Grandstream Networks Expands Interoperability and Reseller Status with Three New Partners Worldwide
Grandstream
Networks expands its interoperability and reseller status with partners worldwide.
Today, September 6, 2011, the company announces three new partnerships:
Vidanetwork Certifies Grandstream for VoIP, Video over IP Applications
Service providers deploying high quality, cost effective voice/video over IP solutions to SMBs and enterprise customers can now seek best value solutions from Vidanetwork and Grandstream. Vidanetwork, a provider of host and premise-based VoIP application software, has certified Grandstream's GXP21xx/14xx series enterprise HD IP Phones and the GXV31xx next-generation IP Video and Multimedia Phones as interoperable with Vidanetwork's Global Office and VOffice Pro platforms. Vidanetwork provides a range of VoIP and video-over-IP applications, including hosted and premise IP-PBX, business trunking and residential broadband services fully integrated into a single VoIP application platform.
http://www.grandstream.com/news/Vidanetwork_Grandstream_PR.pdf
Stonehedge Telecom Achieves Certified Reseller Status from Grandstream
Stonehedge, a single source voice communications and hosted IP PBX provider has added Grandstream IP voice/video telephony equipment portfolio to its suite of product offerings compliments Stonehenge Telecom's hosted IP-PBX solutions. Following extensive testing, certification of Grandstream line of award-winning IP Multimedia and Enterprise IP Phones now brings the opportunity for Stonehedge to further enrich the value and benefits of their hosted PBX solutions to their customers. According to Ralph DiSciullo, president of Stonehenge Telecom, "We opted to pursue a partnership with Grandstream as their R&D, production capabilities and quality really set them apart from the competition."
http://www.grandstream.com/news/stonehenge_grandstream_pr.pdf
Grandstream Now Integrates with SURiX SIP VoIP Door Phone Systems
SURiX, a Latin America manufacturer of Door Phone systems connectable to telephony networks for buildings and gated communities, will begin offering Grandstream SIP Phones with its Door Phone Systems with and without an IP PBX. The combination of SURiX's SIP VoIP Door Phones and Grandstream's IP Multimedia Endpoints offers an excellent low cost solution for Door Phone and IP Telephony integration and provides value added resellers (VARs) and system integrators an innovative, plug-and-play solution for IP based security access & control that's easy to deploy and manage.
http://www.grandstream.com/news/SURiX_PR.pdf
02/08/2011 - CloudTC Announces Glass Interoperability with BroadSoft BroadWorks Voice Application Server
CloudTC announces the interoperability
of the CloudTC Glass 1000 with BroadSoft's BroadWorks
voice application server.
BroadSoft is the leading global provider of software that enables mobile, fixed-line and cable service providers to deliver real-time communications over their IP networks. The firm's BroadWorks platform delivers a broad range of unified communications services including video, voice, hosted call center, conferencing, messaging and mobility, for businesses and consumers worldwide.
The Android-based CloudTC Glass 1000 IP phone has been acclaimed by thought leaders in the industry since its launch in August 2009, and has achieved interoperability with leading infrastructure vendors representing over 50 percent of the IP communications marketplace.
CloudTC is now shipping mass-produced phones to customers in Europe, Asia, and North America. Infrastructure vendors, OEMs, systems integrators, and VoIP service providers have enthusiastically embraced Glass because of the benefits of requiring less time to market, lower investment in R&D, and ability to customize phone features as well as application suites for business users. Because the Glass platform runs on the Android OS, it offers application providers the benefits of an open development environment, flexibility and scalability, and the ability to integrate their applications with unique calling features and a large user screen.

BroadSoft is the leading global provider of software that enables mobile, fixed-line and cable service providers to deliver real-time communications over their IP networks. The firm's BroadWorks platform delivers a broad range of unified communications services including video, voice, hosted call center, conferencing, messaging and mobility, for businesses and consumers worldwide.
The Android-based CloudTC Glass 1000 IP phone has been acclaimed by thought leaders in the industry since its launch in August 2009, and has achieved interoperability with leading infrastructure vendors representing over 50 percent of the IP communications marketplace.
CloudTC is now shipping mass-produced phones to customers in Europe, Asia, and North America. Infrastructure vendors, OEMs, systems integrators, and VoIP service providers have enthusiastically embraced Glass because of the benefits of requiring less time to market, lower investment in R&D, and ability to customize phone features as well as application suites for business users. Because the Glass platform runs on the Android OS, it offers application providers the benefits of an open development environment, flexibility and scalability, and the ability to integrate their applications with unique calling features and a large user screen.
13/10/2010 - Nimbuzz Surpasses 3.65 Billion Mobile Voice Minutes and 150,000,000 Downloads Milestones
Nimbuzz has
surpassed 3.65 billion mobile voice minutes in the last year, and 150,000,000 downloads
of its application since launch in May 2008. With 50% of downloads originating from
the Nimbuzz.com, Nimbuzz, the new-generation mobile service providing free and low-cost
calls and messaging. It is also one of the best performing applications in the popular
Ovi and GetJar application stores.
Ovi Store: Five million downloads since launch in the Ovi store from September 2009 to October 1, 2010, making Nimbuzz a global top five application.
GetJar: 45 million downloads (twice as many as Yahoo! Mobile) on GetJar, and the application is number four of the top five most downloaded global applications.
"Unlike Skype, Nimbuzz is the only open universal communications platform on the market today," said Evert Jaap Lugt, CEO, Nimbuzz. "With 30 million registered users across all major mobile operating systems, our success can be attributed to the fact that Nimbuzz is not solely a chat platform or a VoIP service, but a complete communications service that includes free talk and messaging. We benefit from massive word of mouth with 98% of our users recommend the application to friends, who don't have to own a smartphone to use our product."
Nimbuzz offers free Nimbuzz calls, and low-cost calls through its VoIP product NimbuzzOut, as well as messaging across all major mobile platforms, including Android, iPhone, BlackBerry, Symbian and Java, on over 3,000 handsets.
For the week ending October 1, 2010, Nimbuzz also reported the following downloads in leading app stores of its universal communications platform: "Nimbuzz has been one of the most popular apps for several years running now. It's ease of use, large community and wide handset coverage make it a must have for most mobile users," said Patrick Mork, CMO, GetJar. "We're thrilled to see Nimbuzz doing so well and look forward to seeing them hit the next 50 million downloads."
21/09/2010 - ECG Drives Down Cost of VoIP Expertise
Three obstacles plague VoIP Carriers. First,
the flexibility and complexity of network design options can make network development
take many months or years. Second, support for the complex features of SIP Trunking
and Hosted PBX create a huge burden on staff. Finally, experienced VoIP engineers
and technicians are hard to find, but there are few options for in-depth training.
ECG is changing that. ECG’s experience and speed radically changes the prospects for many VoIP carriers by lowering the costs and delays to deployment, providing access to experts, and enabling staff development.
VoIP networks not only bring amazing flexibility, but also provide unparalleled complexity in network design, quality-of-service, VoIP phone management, Session Border Controller deployments, and reliability. A typical five-person team with a modest VoIP capital expenditure can cost $40,000 per month while they learn and build the system. That home-grown education is tremendously expensive.
ECG shrinks those costs and delays by applying proven best practices for VoIP carriers. Instead of months to deployment, ECG VoIP network design and deployment can take only weeks. One VoIP carrier with services from Michigan to Florida used ECG for just this purpose. ECG analyzed their network and provided detailed assistance improving performance and security.
Once the network is operational, carriers struggle to staff and maintain the network while servicing existing customers. ECG’s solution is to provide on-demand expertise and routine network maintenance. A carrier’s staff can serve existing customers and win new customers, while using ECG for routine maintenance and troubleshooting tasks. A Hosted PBX VoIP carrier based in New York uses ECG for engineering support and technical customer support on a daily basis to free up their staff for hands-on interactions with clients.
VoIP carriers also learn that the skillset for VoIP networks – SBCs, application servers, SIP signaling, QoS Engineering – are not easily found in the job market. Technical staff must be groomed from within. ECG has the only training on industry-standards technology like SIP, MGCP, RTP and session border controllers. A VoIP carrier in Puerto Rico has used ECG’s training to educate Engineering and Operations teams across the company on Network Fundamentals and VoIP Engineering and Troubleshooting.

ECG is changing that. ECG’s experience and speed radically changes the prospects for many VoIP carriers by lowering the costs and delays to deployment, providing access to experts, and enabling staff development.
VoIP networks not only bring amazing flexibility, but also provide unparalleled complexity in network design, quality-of-service, VoIP phone management, Session Border Controller deployments, and reliability. A typical five-person team with a modest VoIP capital expenditure can cost $40,000 per month while they learn and build the system. That home-grown education is tremendously expensive.
ECG shrinks those costs and delays by applying proven best practices for VoIP carriers. Instead of months to deployment, ECG VoIP network design and deployment can take only weeks. One VoIP carrier with services from Michigan to Florida used ECG for just this purpose. ECG analyzed their network and provided detailed assistance improving performance and security.
Once the network is operational, carriers struggle to staff and maintain the network while servicing existing customers. ECG’s solution is to provide on-demand expertise and routine network maintenance. A carrier’s staff can serve existing customers and win new customers, while using ECG for routine maintenance and troubleshooting tasks. A Hosted PBX VoIP carrier based in New York uses ECG for engineering support and technical customer support on a daily basis to free up their staff for hands-on interactions with clients.
VoIP carriers also learn that the skillset for VoIP networks – SBCs, application servers, SIP signaling, QoS Engineering – are not easily found in the job market. Technical staff must be groomed from within. ECG has the only training on industry-standards technology like SIP, MGCP, RTP and session border controllers. A VoIP carrier in Puerto Rico has used ECG’s training to educate Engineering and Operations teams across the company on Network Fundamentals and VoIP Engineering and Troubleshooting.
10/09/2010 - KONNECT Business Phones Certified Interoperable with Xorcom PBX
Aksys
Networks and Xorcom announce that
they have successfully completed interoperability testing between Xorcom’s Complete
PBX IP-PBX models and Aksys’ KONNECT
Business Phones.
Xorcom IP-PBX: Reliable, Flexible Asterisk-based Telephony Platforms
Xorcom’s all-in-one appliances allow seamless communication using both VoIP and traditional telephony protocols. Xorcom’s unique system design ensures the reliability expected from a proprietary telephony system, while providing the flexibility, scalability, easy integration and competitive pricing afforded by an open source platform. The Xorcom IP-PBX line supports all the features of the extremely popular Asterisk PBX operating system while providing additional benefits, such as dual-PBX hot failover with the award-winning TwinStar feature, high density of telephony interface combinations in a small footprint, etc. The Complete PBX, sold and supported by Xorcom in the United States and Canada, features a customized user interface to streamline the implementation and enhance the functionality of the telephony system.
KONNECT Business Phones provide the simplest end-user experience in the market and are also simple to install and manage.
The KONNECT Business Phones are able to deliver the most intuitive, flexible and responsive user experience in the market. Features like visual voicemail, call forward, call record, and call park, provide simple and clear prompts and soft-keys to ensure the user is able to make full use of these features with no confusion and minimal training. For the integrator or installer, these features are pre-loaded on the Xorcom Complete PBX, as are the automated provisioning tools which ensure installation and configuration can be done smoothly and quickly for any size of deployment.
KONNECT Business Phones are fully SIP compliant and are simple and easy to install and manage. This results in happier end-users who need less training to be fully productive on their new phone system. KONNECT Business Phones ensure that the integrator can make full use of the capabilities of the Xorcom product by presenting access to features and functions in a more friendly and accessible manner.
Xorcom and KONNECT Business Phones deliver a simple, state of the art communications solution.
The fit between these two solutions is ideal in that both products are designed to simply and easily deliver the full potential of Open Source Telephony for the end-user and the integrator alike. The Xorcom Complete PBX and the KONNECT Business Phone handsets make it simpler to get full value of all the features and cost savings available from an Asterisk-based solution.
11/08/2010 - FreedomVoice Not in Any Danger of Going Bankrupt, Cites Name Confusion as Source of Rumors
FreedomVoice announces
to customers and partners that rumor of a bankruptcy stems from confusion with another
telecom company of a similar name. In early August, a company by the name of Freedom
Communications USA filed for bankruptcy in Tennessee. The likeness of this company’s
name has led many in the industry to incorrectly attribute this event to FreedomVoice,
prompting concerned phone calls and emails. Officers at FreedomVoice say nothing could
be further from the truth.
“We’re headed in exactly the opposite direction,” says Eric Thomas, President and CEO of FreedomVoice. “Over the past couple of years, we’ve experienced tremendous growth with our flagship product, the FreedomIQ Hosted VoIP PBX. As our technology continues to attract and impress SMBs, we find ourselves in a great position with a well-received product in hand.”
While the full names are similar, the real confusion is in how the two companies are referred to in short.
“Many in the industry choose to refer to us simply as ‘Freedom,’” says Thomas. “Freedom Communications USA apparently experiences the same name-shortening, so it’s easy to see how there might be a mix-up. It’s a shame to have to come out and make a statement in the wake of another’s misfortune, but we felt a strong need to clear up this confusion and reassure our customers and partners that we’re doing better than ever.”
Both telecom companies emerged after the passing of the FCC’s Telecommunications Act of 1996, but that’s where the similarities end:
- Freedom Communications USA is based in Dickson, Tennessee, while FreedomVoice is based in Encinitas, California.
- Freedom Communications USA operates in 10 states, while FreedomVoice does business nationwide.
- Freedom Communications USA targets residential and business customers with phone, data, satellite, and wireless services, while FreedomVoice specializes exclusively in advanced solutions for the SMB market.
“The people contacting us are shocked,” explains Thomas. “They’re the first to sing our praises, and in disbelief that we could be having problems. We get the privilege of telling them, ‘No, we’re not going anywhere but up.’”
This is not the only time in the recent past that the name ‘Freedom’ has caused confusion for FreedomVoice over bankruptcy filings. A little less than a year ago, a large media company by the name of Freedom Communications, Inc., also referred to as ‘Freedom’ for short, filed for bankruptcy in Delaware.
05/08/2010 - Ooma Introduces New International Calling Plans
Ooma announces
three new international calling plans that further extend the cost savings and convenience
of calling friends and family around the world. With these plans Ooma customers can
now call 70 countries for less than a penny per minute and call other countries at
reduced “bulk discount” rates. With the introduction of these new plans and the recently
announced Ooma Mobile Application, Ooma offers the most affordable and functional
international calling experience.
Ooma offers its customers a comprehensive international calling program that affords multiple ways for consumers to save on international calls:
Ooma International Calling Bundle
Available today, Ooma customers can select from three new plans that best fit their calling lifestyle. The first plan offers Ooma customers 1,000 free minutes of talking to friends and family in 70 countries for the low rate of $9.99 per month. The second plan, which also costs $9.99 per month, offers access to significantly reduced “bulk discount” rates for use when calling all other international countries. The third plan, which is available only to Ooma Premier customers, provides both the 1,000 free minutes to call 70 countries and the access to “bulk discount” rates for calling other international countries combined for $9.99 per month. The original Ooma International calling bundle – $4.99 a month for 500 minutes of calling to 70 countries – also remains available for Premier subscribers. For more information, visit the international calling page: http://www.ooma.com/products/international-rates
Ooma Mobile Application
Save mobile minutes and up to 90 percent on international calls compared to traditional mobile calling plans with the Ooma Mobile App – now available for iPhone, iPod Touch and iPad – by placing calls over any Wi-Fi connection or 3G network. For more details, visit the Ooma page in the App Store: http://itunes.apple.com/us/app/ooma-mobile-for-iphone/id348758102?mt=8
Ooma to Ooma Calling
Placing Ooma to Ooma calls is always free. If you own an Ooma system and send an Ooma system to friends and family abroad, they can assign a U.S. phone number to that system and call back to the states for free.
20/07/2010 - VoxOx Launches Local and Custom Phone Numbers in U.S. and Canada
VoxOx announces
the immediate availability of virtual phone numbers. This new offering enables consumers
to create multiple customized telephone numbers that route to their VoxOx account,
and can also be routed to other phones (work, home, mobile, etc.). Consumers can select
their virtual phone numbers from more than 100 area codes in the U.S. and Canada.
They also have the ability to do a “vanity” search for memorable digits and word spellings.
VoxOx virtual numbers are being offered in addition to the free phone numbers that every VoxOx user receives upon sign-up. Each virtual number costs $1.95 per month or $19.95 per year – one third of the cost of Skype’s online numbers. Differentiators of VoxOx virtual numbers include support for inbound SMS and fax.
VoxOx also announced the addition of free global numbers, made possible through an alliance with the iNum initiative. iNum is a special global area code (+883) dedicated to enabling new IP communications services. These global numbers make it easier for friends, family and colleagues overseas to dial VoxOx users from landlines and mobile phones, and other VoIP networks worldwide at local rates or free. Once a VoxOx user activates the free iNum number, the person’s overseas friends can dial him or her from any phone via one of the local access numbers in 45 countries as a two-stage call. As a result, the VoxOx user pays nothing to receive the international call and the overseas friend only pays for the local call to the access number.
With these new phone number options, VoxOx provides some of the best flexibility in the industry for choosing a number that meets a user’s objectives. Currently, consumers who sign up for VoxOx in all supported countries receive:
- A free randomly selected U.S. phone number and initial calling credit
- An option to add multiple virtual phone numbers, customizable by area code and “vanity” digits
- A free global iNum number to enable overseas friends to call VoxOx users at local rates or free
- The option to forward a consumer’s existing cell phone number to VoxOx without having to give out a new phone number
24/06/2010 - D2 Technologies Adds Full Skype Support to Its mCUE IP Communications Interface for Android-based Devices
D2
Technologies announces that its mCUE product line now fully supports Skype’s SkypeKit
SDK. mCUE gives OEMs and ODMs a quick and economical way to speed Android OS-based
devices to market that offer a communications client with native Skype interoperability
and powerful integrated communication capabilities. Combining the SkypeKit SDK with
mCUE’s advanced user interface and high quality media engine offers customers the
added benefit of mCUE’s advanced multi-mode, multi-session, multi-protocol engine
for full interoperability with multiple carrier and enterprise communications systems
and other social networking services.
D2’s patented mCUE is an innovative VoIP and IP communications client that provides users with advanced presence-based and push-to-x control of circuit switched PSTN/cellular and VoIP calls, video chat/call, IM, presence, email, SMS, PBX, and other features typically only available on PC-based unified communications soft clients. Its revolutionary presence-based user interface, built on top of a multi-identity, multi-session, multi-protocol engine, enables flexible interoperability with multiple communications services and, in addition to Skype, social networking services like GoogleTalk, Yahoo!, MSN and AIM. mCUE also provides multi-radio, multi-network seamless voice call handover to provide call cost and voice quality optimization.
26/05/2010 - PortaOne Announces Interoperability of PortaSwitch with TelcoBridges Tmedia Gateways
PortaOne announces
a new partnership with TelcoBridges following
successful interoperability testing between Tmedia products and PortaOne's PortaSwitch
VoIP call control and billing management software platform. The announcement, made
at International Telecommunications Week 2010, assures telecom service providers not
only of seamless operation between the products, but also a combination of reliability,
scalability and system flexibility suitable for today's most demanding network and
market requirements.
A recognized leader in the design and manufacture of telecommunications hardware, TelcoBridges has developed Tmedia, the industry's highest density VoIP gateways that deliver exceptional performance, energy efficiency and interoperability. Based on a unified technical architecture, each Tmedia VoIP/media gateway integrates T1/E1/J1, DS-3 and OC3/STM-1 connectivity and multiple concurrent signaling protocols such as SS7, ISDN, SIP, SIGTRAN, and H.248 in the same unique device. Carriers and service providers in over 50 countries use TelcoBridges' VoIP gateways to drive network convergence, consolidate multiple disparate devices, and more cost-effectively expand their network footprints.
By integrating with PortaSwitch, Tmedia gateways ensure the highest levels of performance are maintained. PortaSwitch, a Class 5-level product, is able to handle the full range of call control and billing functions of a diversified digital services company, from billing and provisioning to advanced call features, monitoring and reporting. The comprehensive product consists of a real-time billing system, class 4/5 SIP softswitch, and application servers that deliver converged VoIP billing and provisioning, SIP call control, unified messaging, IP Centrex and hosted IP PBX, callback management, IVRs, conferencing and more.
Tmedia gateways offer a high system density for systems with limited rack availability. In a single 1U or 2U gateway, Tmedia supports up to 2048 voice channels; it also reduces SS7 costs by having multiple point codes in a single box, or multiple boxes with a single SS7 point code. The product's non-blocking capabilities ensure that multiple signal protocols, as well as multiple codecs, can be run at the same time.
To both lower operating costs and provide a greener infrastructure, Tmedia gateways run on up to 2/3 less power than other gateway devices of similar capacity. Power, cooling and data center co-location costs are all reduced, thanks to the green design.
19/04/2010 - CounterPath Releases Network-Based Mobile Enterprise Convergence Mashup Application
CounterPath announces
the launch of NomadicPBX a turnkey platform for enabling converged mobile and broadband
SIP voice, messaging and presence services. Available immediately, NomadicPBX gives
wireless carriers and mobile virtual network operators a cost-effective way to launch
enterprise fixed-mobile convergence services.
The NomadicPBX application is a presence-based, fixed and mobile voice, and instant messaging/short message service technology mashup with a select set of enterprise-ready features. Based on CounterPath’s Network Convergence Gateway platform and leveraging the company’s patented technology, NomadicPBX enables wireless operators and other service providers to extend their feature sets into small and medium enterprises, which currently account for one-third of all hosted VoIP deployments, according to a recent Infonetics Research survey.
The NomadicPBX’s network-based features let operators push deeper into the business market, creating new revenue streams and differentiating themselves based on innovative services rather than price alone. Carrier customers that currently deploy CounterPath’s NCG platform and Bria client can use the NomadicPBX configuration to create services relevant to other market segments, too.
NomadicPBX uses presence for status updates and real-time management of call and message delivery across a wide variety of endpoint types, from PCs to feature phones to smartphones, all from any vendor. This flexibility means enterprises can integrate standard mobile network services into their enterprise communications architecture by leveraging their existing devices. End users benefit from a single number and identity, which lets them be reached immediately from any mobile, desk phone or VoIP softphone, including a client running on a mobile phone. This flexibility allows enterprises to implement more efficient, cost-effective and responsive communications within their organizations and with customers and business partners.
NomadicPBX also:
- Enables convenient extension dialing such as short-dialing or speed calling from any mobile handset
- Supports core calling features found in most PBX solutions
- Operates completely independent of the enterprise’s IT infrastructure, eliminating CIO and IT manager security and support concerns
- Improves productivity and lowers costs for mobile workforces
View a video demonstration of NomadicPBX at http://counterpath.com/nomadicpbx.
19/04/2010 - Ooma Announces PureVoice Technology
Ooma announces
the availability of Ooma PureVoice technology, designed to ensure excellent voice
quality under the most demanding broadband conditions.
Ooma PureVoice technology is comprised of four key components:
Advanced voice compression
Ooma uses an advanced voice compression algorithm that reduces bandwidth consumption by 60% over standard VoIP technology and is more capable of withstanding packet loss without degradation. This leaves you with more bandwidth for all other online activities and increases the likelihood that your voice traffic will be delivered properly by your ISP.
Wire-speed QoS
Even though Ooma uses only a fraction of the bandwidth of standard VoIP technology, preserving voice quality requires that those packets arrive on time. The Ooma Telo prioritizes voice packets without slowing down the rest of your network. This way you can enjoy crystal clear calls even as you are uploading your latest video clips.
Adaptive redundancy
Packet loss is the enemy of VoIP - it can cause voice to sound stuttered or garbled. The Ooma Telo detects packet loss on your Internet connection and automatically sends redundant packets to boost the clarity of your phone call.
Encrypted calls
Ooma takes your privacy seriously. We use the same encryption technology to protect your conversations that governments use to protect classified data. This makes Ooma even more secure than the traditional landline.
For more information and to listen to the differences between Ooma with PureVoice Technology and other services, go to: www.ooma.com/purevoice.
Ooma has also announced the availability of new home smart phone features including Google Voice Extensions and Ooma Voicemail Transcriptions.
Google Voice Extensions
Ooma simplifies and enhances the Google Voice user experience, enabling consumers to take advantage of the complementary capabilities found in both offerings for a truly integrated and seamless phone experience. Google Voice users can integrate the Call Presentation, Listen In, and caller-ID features with their Ooma system as well as access Google Voice voicemail at a touch of a button. Now, consumers can enjoy one voicemail box for all their phone messages and present one caller-ID to all callers.
Ooma Voicemail Transcriptions
This service converts Ooma voicemail messages into text and delivers it to a mobile phone or email account associated with the Ooma profile. Benefits of Ooma Voice-to-Text:
- Read voicemail wherever there’s email access — on a mobile phone, portable device or computer
- Quickly see who called and what they called about
- No need to write down phone number or directions — it's all there in your inbox
- Choose to respond to the voicemail through email by just forwarding your response to the caller
06/04/2010 - Business Jet VoIP has Landed Thanks to Dassault Falcon
Dassault Falcon delivered the first Falcon business jet (Falcon 7X, s/n 85) with Honeywell's new MCS 7120 Swift Broadband Communications Gateway. The product provides a fully integrated wired and wireless cabin communication system and high speed global connectivity via the Inmarsat I4 Satellite Network. The aircraft is the first business jet in the industry capable of providing fully-managed end-to-end VoIP telephony services over the Swift Broadband network. Managed VoIP services deliver significantly higher quality audio performance because of dedicated bandwidth to each call, assuring excellent audio fidelity.

24/03/2010 - VoIP Supply Launches Enhanced HD VoIP Category
VoIP
Supply announces the launch of an enhanced category for HD VoIP technology. The
new enhanced HD VoIP category features educational information, manufacturer solution
overviews and a full suite of HD capable handsets.
“High definition voice calling is no longer the future; it’s the present” stated Garrett Smith, Director of Marketing and Business Development at VoIP Supply. “With most major VoIP manufacturers and service providers supporting HD voice calling, now is the perfect time for businesses of all sizes to learn about the benefits the technology delivers and put it to use within their operations.”
VoIP Supply’s HD VoIP category features everything you need to know about HD voice calling including educational articles like: These educational HD articles are accompanied by HD solution overviews from some of today’s leading VoIP manufacturers such as Aastra, AudioCodes, Cisco and Polycom. In addition, the HD VoIP category features today’s most popular HD capable handsets for those looking to take advantage of HD voice calling today.
18/03/2010 - Acme Packet Rolls Out HD Voice
Acme
Packet announces HD Voice - Xtended Reach, a set of capabilities for its industry-leading
Net-Net session border controllers family that bridge HD voice services and applications
across IP network borders and ease the transition from standard definition to HD voice.
Support for new HD coder/decoders on Acme Packet’s Net-Net 9200 platform delivers
the transcoding and transrating flexibility needed by fixed and mobile service providers,
as well as enterprises and contact centers, to leverage the new generation of codecs
found in audio endpoints such as HD-capable mobile handsets, HD telepresence, and
IP phones used in audio conferencing and contact center solutions. Additionally, new
codec management functions for Acme Packet’s entire Net-Net SBC family control codec
selection and session routing based on codec to optimize subscriber quality of experience.
HD voice features endpoints equipped with wideband coders/decoders that deliver CD-quality audio, enabling a much richer communications experience for both wireless and wireline services. Communications-oriented business functions such as audio and video conferencing and customer service can be significantly enhanced by the life-like clarity of HD voice. The high audio quality levels also improve perception in challenging environments such as international phone conversations and noisy venues. Mobile service providers also view HD voice as a driver for the adoption of fixed mobile substitution, as the single-pair wiring still found in many homes is insufficient for supporting HD phones, many of which require Category 4 wiring or better.
In spite of its promise, delivery of HD voice across IP network borders can be challenging. First, wireline, 3G GSM/UMTS, CDMA and emerging 4G LTE networks each utilize different wideband codec standards. Second, IP networks that support wideband codecs must still be able to communicate with those that do not. Third, calls originated on HD-capable networks are sometimes routed across transit networks that are not HD-capable, leading to degraded voice quality. To address these challenges for VoIP service providers, enterprises and contact centers, Acme Packet’s Net-Net 9200 SBC supports transcoding and transrating for three new codecs:
- G.722 – a wideband codec used in wireline HD VoIP services and applications
- G.722.2 – also known as Adaptive Multirate Wideband (AMR-WB), is the standard codec used for GSM/UMTS-based HD voice services
- EVRC-B – Enhanced Variable Rate Codec Revision B (EVRC-B), a bandwidth-efficient narrowband codec used in CDMA services
Acme Packet’s HDV-XR, which leverages Net-Net OS across Acme Packet’s Net-Net SBC family, delivers additional features that further assure HD-quality VoIP service across network borders:
- Codec re-ordering elevates wideband codecs to the top of codec preference lists in the Session Description Protocolused to set-up of SIP calls. For scenarios where endpoints use the same HD codec, this ensures transcoder-free operationand preserves HD voice communications end-to-end.
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The Net-Net SBC, as well as Acme Packet’s session routing proxy, the Net-Net Session
Router, can route SIP sessions based on the selected codec. If an endpoint uses a
wideband codec, the Net-Net SBC or SRP will attempt to route that session over a transit
network that ensures it will not be transcoded to a lower quality codec. If the session
negotiates the use of a narrowband codec, it is routed along an alternate path configured
for standard definition service.
25/02/2010 - VoIPon Joins Xorcom Distribution Channel
Xorcom announces
that VoIPon has signed an agreement
to distribute the entire Xorcom product range to qualified resellers. The full Xorcom
line broadens VoIPon's reach into the open source SMB hybrid PSTN + VoIP phone system
space.
USB 2.0-connected Channel Banks Provide Additional Ports, Effortlessly
Astribank is a versatile and powerful channel bank that supports all the common telephony lines and trunks: FXS, FXO, BRI, PRI and R2. Specifically designed for the Asterisk IP-PBX, the Astribank driver is a part of the standard Asterisk distribution, making its integration into any Asterisk-based PBX effortless.
Xorcom IP-PBX: Reliable, Flexible Asterisk-based Telephony Platforms
Xorcom's all-in-one appliances allow seamless communication using both VoIP and traditional telephony protocols, such as FXS, FXO, BRI, T1/E1 PRI, T1 CAS and E1 R2. Members of the series differ in the number of supported users, starting from the basic XR1000, which is suitable for SOHO, up to the robust XR3000, which supports up to eight PRI connections along with hundreds of analog and IP extensions. The award-winning TwinStar feature enables dual-server redundancy for the complete PBX, including all telephony trunks and interfaces, as well as IP phones.


USB 2.0-connected Channel Banks Provide Additional Ports, Effortlessly
Astribank is a versatile and powerful channel bank that supports all the common telephony lines and trunks: FXS, FXO, BRI, PRI and R2. Specifically designed for the Asterisk IP-PBX, the Astribank driver is a part of the standard Asterisk distribution, making its integration into any Asterisk-based PBX effortless.
Xorcom IP-PBX: Reliable, Flexible Asterisk-based Telephony Platforms
Xorcom's all-in-one appliances allow seamless communication using both VoIP and traditional telephony protocols, such as FXS, FXO, BRI, T1/E1 PRI, T1 CAS and E1 R2. Members of the series differ in the number of supported users, starting from the basic XR1000, which is suitable for SOHO, up to the robust XR3000, which supports up to eight PRI connections along with hundreds of analog and IP extensions. The award-winning TwinStar feature enables dual-server redundancy for the complete PBX, including all telephony trunks and interfaces, as well as IP phones.
16/02/2010 - Vopium Adds Instant Messaging with Free calls to Skype and Gtalk Users
Vopium has
announced a range of new features to its award winning Mobile Application. Now it's
possible to make FREE calls to Skype, Gtalk and Vopium users when they are online.
Besides chatting with friends on Skype, MSN, Yahoo! Google Talk, AIM, and ICQ, one
can also follow them on Twitter.
The new iPhone and iPod App also includes a number of cool new features. For example it is easier to send text messages and check and transfer balance to other users directly from the iPhone.
Instant Messaging is now available on Android, Blackberry, iPhone & iPod and Windows Mobile and will be released for Symbian and Java in the coming days.
Improved SMS, own VOIP client and introduction of web based calls and IM.
Already covering Symbian and Windows Mobile, the popular seamless SMS service will be available on Android & Blackberry soon. Vopium automatically detects when an SMS is being sent abroad, enabling users to send SMS normally and smoothly without changing the way the messages are sent, ensuring users great savings on international SMS sent via Vopium without any hassle.
Vopium also now has its own VoIP client, allowing Wi-Fi calls and 3G VoIP calls on devices without an embedded VoIP client, allowing Vopium users to make VoIP calls on handsets like e.g. Nokia N85, N97 and Nokia 5800 Xpress Music.
Introduction of web based calls and IM makes it very easy to access and call your contacts everywhere in the world, avoiding expensive roaming charges. Users can log on to any PC/Web browser from anywhere in the world and access their mobile and IM contacts. Users can also receive incoming calls from other Vopium users 100% free.
15/02/2010 - Voxbone Introduces SMS Support for iNum
The move marks a breakthrough in iNum usage, as wireless subscribers from a growing number of prominent carriers – including Vodafone, T-Mobile, Orange, Virgin, and Boost Mobile – now are able to text these numbers. The service is already available in the United Kingdom, France and the United States, at prices ranging from 10 to 20 pence per message in the U.K., for example. Voxbone will be adding reachability from more wireless carriers in more countries in the coming weeks.
iNum numbers have a prefix of +883, the ITU-assigned international code for the Internet, just as +44 is the code for the U.K. and +1 refers to the U.S. As a wholesaler of direct-inward-dial numbers and IP transport provider, Voxbone receives calls – and now SMS messages – to numbers with this code and delivers them over IP to its carrier customers, for delivery to their end users.
11/02/2010 - VoicePulse and YouMail Partner to Create an Enhanced Voicemail Experience
VoicePulse announces
the integration of YouMail, the premier mobile consumer voicemail service with VoicePulse
residential calling plans. Residential users now have the option of converting their
VoicePulse voicemail to YouMail services. Customers can take advantage of YouMail’s
enhanced voicemail feature set while still enjoying VoicePulse premiere residential
calling services.
VoicePulse with YouMail provides customers with more control over voicemail with features such as:
- Thousands of custom greetings and the ability upload your own greetings
- Voicemails received via your phone, email or through the YouMail account center
- Manage your voicemail account with Blackberry, Android and the iPhone apps
- The ability to upgrade to premium services such as transcriptions and additional voicemail storage
VoicePulse broadband phone service plans start at $14.99 per month and include over 20 features such as:
- Telemarketer Blocking, Virtual Numbers, Anonymous Call Rejection, Call Transfer and Call Hunt
- Customizable CallerID with a Name and CallerID Block
- Voicemail with optional e-mail delivery of messages as sound attachments
- Free Local Number Portability
- Online Account Center management
02/02/2010 - Tpad Expands Consumer VoIP Convenience with Google Checkout
Tpad announces
the addition of Google Checkout to their roster of online payment options for their
consumer services division. Tpad is dedicated to providing their customers in 28 countries
with low-cost, superior service call packages, and the addition of this payment option
further enhances the convenience of purchasing their VOIP call credits online.
Google Checkout has become a strong contender in online payment processors, allowing secure transactions for consumers integrated with fraud protection. The benefits for Tpad and its customers are numerous, starting with an increased level of security - via this partnership with Google Checkout they can purchase the services they need at each billing period without entering their information each time.
Tpad’s strategy is also in direct response to the need to differentiate their brand from the competition in the residential VoIP marketplace by addressing the needs of consumers who shop frequently online and want to further enhance the security level of their transactions. Using Google Checkout allows them to purchase their services from Tpad in addition to thousands of other merchants without exposing their credit card information. In addition, customers can keep track of their online purchases from Tpad and their other Google Checkout transactions as well.
Tpad has made it easy to access these savings; customers simply register for a free Tpad VoIP account. Once you verify your email, log in with your 7 digit Tpad number and password. You can select the level of credits you want to purchase and when you click the Google Checkout button you will be taken to the Google secure website to complete your purchase.
Customers can use any standard mobile phone with free Nimbuzz or Fring SIP software installed to access the Tpad VoIP network or download a free PC soft phone such as ZolPer, Express Talk or Xlite. If customers have a SIP or VoIP compatible device such as an IP phone, Google Android phone, Apple i-Phone, Nokia Wi-Fi Mobile or Linksys ATA , they can access the service as well. Consumers and Tpad both benefit from this payment option; for consumers they can purchase just the amount they need without any hidden transaction fees or being tied to a long-term commitment and Tpad is able to further enhance its level of customer care.
The Tpad VoIP advantage is not only in the savings, but in the level of crystal clear voice quality and superior customer care from Tpad’s tier one data center. Unlike the case of making an international call on many mobile carriers, the receiver of the call is not charged when the connection is made, no matter where they are on the globe or the length of the call.
20/01/2010 - VoIP Supply Finds Fax over IP Success with FaxxBochs
VoIP
Supply announces the successful introduction of RockBoch’s
FaxxBochs Fax over IP solution to it’s base of over 75,000 customers. FaxxBochs,
which combines a low cost Fax over IP gateway with an affordable monthly service,
allows businesses of all sizes to make 100% reliable analog to analog faxing over
an IP network possible.
“As more and more businesses migrate their business communications systems from analog to IP there is a great need for a reliable fax over IP solution,” stated Garrett Smith, Director of Marketing and Business Development at VoIP Supply, LLC. “After a few disappointments with other Fax over IP solutions, VoIP Supply finally found a winner in FaxxBochs. All of our initial customer deployments have been met with great success and we’ve even integrated the solution in our own operations.”
"Reliable fax over IP has been an ongoing problem, especially in the satellite space,” stated RockBochs President Chad Behling. “With the FaxxBochs solution, we are now able to provide reliable fax performance over ANY network, regardless of latency or jitter."
FaxxBochs is the world’s only 100% reliable analog-to-analog fax over IP solution. While e-fax has become extremely popular over the past few years, it does NOT address several fundamental flaws. E-fax is NOT secure, period. Emailing sensitive documents to an e-fax provider traverses the public internet, increasing the likelihood of interception. Digital signatures are cumbersome and impractical. User training becomes extremely complex in larger organizations.
With FaxxBochs, all fax transmission is encrypted via 256-bit AES encryption between the FaxxBochs CPE and the FaxxBochs Datacenter. You can continue to use your standard fax machine, no special equipment required. With FaxxBochs, nothing changes for the user, they simply continue to use the fax machine as they always have. FaxxBochs offers a full archival of all inbound and outbound faxes available. FaxxBochs also offers Fax-to-Email available for inbound faxes. FaxxBochs includes an intuitive, web-based interface to view and manage all inbound and outbound faxes.
Prices for the FaxxBochs Fax over IP gateway start at $175 USD with the corresponding monthly service starting at just $30 USD per month. For more information about the FaxxBochs Fax over IP solution or to sign-up, please visit .






