VoicePulse announces interoperability with IPitomy's Pure IP PBX platform. IPitomy designs and manufactures a complete line of IP telephony equipment with an advanced feature set for businesses. IPitomy's PBX combined with VoicePulse's SIP origination and termination services create a complete phone solution for businesses of all sizes.

VoicePulse can now provide new customers using IPitomy's IP PBX with official configuration guides when setting up VoicePulse VoIP services on their PBX. Businesses using IP PBXs such as Digium's Asterisk, AsteriskNow, Fonality's PBXtra, trixbox, FreePBX, FreeSwitch, Switchvox, and now IPitomy can benefit from VoicePulse's competitive international rates, toll-free services, and failover features all on a "Tier 1" back bone network.

Despite lingering questions about overall quality of service, enterprises are becoming more comfortable with the use of SIP trunking as a low-cost alternative to legacy telecom services, a change that will result in a growth in deployment from 8.5 million SIP trunks in 2009 to 24.3 million trunks in 2013, according to the latest report published by Heavy Reading IP Services Insider, a subscription research service from TechWeb's Heavy Reading.

XO Communication announces the availability of a new Savings Estimator tool that gives enterprises the ability to calculate an approximate cost-savings benefit of utilizing XO Enterprise SIP. Designed specifically for multi-location enterprises, the award-winning XO Enterprise SIP enables customers to simplify, streamline and achieve better cost savings by transforming their distributed voice network architecture to a more centralized and cost-effective VoIP solution.

The XO Enterprise SIP Savings Estimator tool provides a snapshot of potential savings by factoring in the number of employees, network locations and intra-company long distance calls. The tool also takes into consideration the multiple cost-savings benefits of XO Enterprise SIP, including increased network management efficiency and lowered operating costs as a result of reducing equipment, local voice trunks, long distance and multiple voice and data network charges.

Unified Communications is coming of age and it's being driven by the urgent demand for SIP trunking, which lowers operating costs and delivers rapid ROI. To address the need for information on the what, why and how of Unified Communications and SIP trunking, Ingate Systems is partnering with TMC, thought-leaders and vendors in the space to present the new SIP Trunk-Unified Communications Summit at ITEXPO West 2010.

Free to all ITEXPO attendees, the Summit is a three-day seminar series providing in-depth educational information on SIP trunking and Unified Communications. The Summit will include general information panels and technical insight sessions from both the service provider and enterprise perspectives, and will feature visionaries driving the industry.

Virtual PBX has expanded its Open VoIP Peering service to include support of any SIP-compliant phones. Users of any SIP-compliant softphone or desktop phone can now tap into the benefits of the Virtual PBX Open VoIP Peering service. This service allows small businesses to take advantage of the robustness and features of traditional hosted PBX services and leverage the cost-effectiveness of VoIP technology. Today’s announcement allows customers more freedom to mix and match SIP-compliant phones with Virtual PBX’s hosted PBX services.

Global Crossing is providing its suite of SIP Trunking services to Clarity Consulting for its hosted contact center management product. Global Crossing's SIP Trunking services enable Clarity Consulting and its partners to deliver a new customer service center software solution that enhances the call control functions of Microsoft Office Communications Server 2007 Release 2 for enterprises worldwide. Global Crossing was one of the first SIP trunking service providers qualified with Office Communications Server 2007 R2.

Cable provider KMS has transferred all Residential VoIP Telephony services to the DIGITALK NGN SIP Application Server. KMS have been operating a residential primary line replacement service using Open SER, an open source technology for its voice service as part of its cable TV, internet and voice packages. However, the limitations of features and operational expenses of integration, maintenance and management have proved a costly overhead.

The integration of the DIGITALK SIP Application server allows KMS to offer a full class 5 feature set with DIGITALK’s Broadband Telephony application and consolidate all management and maintenance onto a single platform footprint, including DIGITALK’s integral Session Border Controller.

VoX Communications has completed development of a downloadable mobile VoIP application for use on Symbian mobile phones.

According to Gartner, 80 million Symbian devices were sold worldwide in 2009, or approximately 47% of the entire global smartphone market. Symbian devices are so dominant in the marketplace today, that it would take the combined sales of the iPhone, RIM, Windows Mobile and Android to equal the same volume opportunity. VoX estimates that its app can be downloaded to more than 100 million Symbian mobile phones that are currently in use. Gartner also projects that Nokia will sell in excess of 100 million Symbian phones in each of the next 3 years.

UM Labs announces support for Skype for SIP. This means that Organisations can now use the UM Labs SIP Controller to simplify the task of linking their corporate PBX to the Skype for SIP service. Skype for SIP offers a number of compelling benefits, including free inbound calls from any Skype user and low cost local numbers for inbound and outbound calls in over 25 countries. A Skype for SIP link requires that the corporate PBX accepts inbound SIP calls over the Internet. The UM Labs SIP Security controller monitors these calls protecting the corporate PBX from a wide range of security threats, many of which are not blocked by a standard Firewall.

XO Communications in collaboration with TMCNet and BroadSoft, will host an interactive webinar on May 12th entitled, "Beyond SIP Trunking - Unify the Enterprise." The event will bring together several SIP trunking experts to discuss the latest industry trends and insights on how distributed enterprises can streamline their managed voice services.

Specifically, the hour-long session will cover topics such as the benefits of SIP trunking services and future trends, how to evaluate the return-on-investment of SIP, as well as things to consider when evaluating SIP trunk service providers. It will also offer an inside glimpse of various enterprise networks before and after SIP trunking was implemented.

Panasonic announces Digium has certified the Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for use with the Asterisk telephony platform. By leveraging Panasonic's market leadership in DECT cordless telephones and Digium's innovative open source PBX, this alliance is a winning combination.

The Panasonic TGP500 series of SIP Cordless Phone Systems boasts outstanding voice quality and a range of productivity-boosting options in a feature-rich desk phone replacement. DECT 6.0 ensures no interference with wireless networks, and the convenient cordless design eliminates the need to run dedicated network wiring to each employee workstation. It is ideal for home and business environments.

Acme Packet announces the launch of the Acme Packet enterprise seminar series, Deploying SIP Services Successfully, kicking off in San Francisco on May 20, continuing on to New York, May 25, Chicago, May 26 and ending in Washington, D.C., on June 2. The goal of this series is to educate enterprises, contact centers and government agencies about the significant challenges of integrating SIP trunking, video conferencing, unified communications, contact centers and hosted cloud communication services, and the role of session border control in ensuring interoperability, security, quality, availability, and regulatory compliance.

Broadvox and UM Labs announce interoperability certification between Broadvox GO! SIP Trunking and UM Labs’ family of enhanced SIP security controllers.

“Interoperability certification provides resellers and customers with confidence that Broadvox’s growing ecosystem of technology partners can support their specific PBX environment and provide enhanced cost effective connectivity and security solutions”, said Isaac Parampottil, Director of Product Management at Broadvox. “We welcome UM Labs as a technology interop partner.”

Digium and Yealink announce an interoperability partnership and certification of the Yealink SIP-T2x series IP phones. Yealink SIP-T2x series are high performance and affordable SIP telephones that help businesses leverage the increasing benefits of VoIP telephone systems. Yealink SIP phones provide high quality audio, a broad range of voice codecs, security protection for privacy, and rich telephony features. Yealink enterprise HD IP phones now are compliance-tested by Digium for interoperability with Asterisk.

The SIP Forum has announced that its next SIP Interoperability Testing event, SIPit 26, will be held May 17-21, 2010 in Kista, Sweden. The event will be hosted by Edvina and TANDBERG, and sponsored by Intertex, Ingate, .se, and the IPv6 Forum as an association sponsor.

Conducted by the SIP Forum twice a year, SIPits are the world’s premier interoperability testing events for the SIP, bringing together the leading SIP application developers, service providers and IP communications equipment manufacturers to ensure their SIP implementations work seamlessly together in an IP network testing environment. An important goal of the SIPit events is to help refine both the SIP protocol and its implementations in order to further establish SIP as a global interoperable standard for real-time Internet communication services.