Acrobits announces the long awaited release of their popular business caliber SIP client Groundwire on Android. Groundwire includes all the features of Acrobits Softphone as well as the additional features business users need. Including but not limited to transfer and attended transfer, call conferencing, multi line and voicemail notification; Groundwire puts all the tools professional SIP users need in the palm of your hand.

With Groundwire, Acrobits also brings support for ZRTP to Android. The most advanced method for call encryption in VoIP, ZRTP is a must have for users who want the most secure calls possible. In addition, Acrobits adds support for SDES SRTP. Both features will be available in both Acrobits Softphone and Groundwire for Android. Groundwire is available on Google Play and the Amazon Marketplace now. A new update for Acrobits Softphone is also available which adds support for ZRTP and SDES SRTP.

Digium introduces the G100 and G200, the first in a family of cost-effective VoIP gateways that simplify the process of deploying converged media networks. Built on a powerful combination of the Asterisk open source communications engine and a state-of-the-art embedded platform, the new gateways provide the best value for Asterisk communications solutions.

Digium’s gateways are built to support both TDM-to-SIP and SIP-to-TDM applications. In a TDM-to-SIP deployment, the gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.

Dialogic announces that SIPxchange chose Dialogic as its core technology provider. The Dialogic ControlSwitch System and media gateway solutions provide SIPxchange with robust routing capabilities, while supporting both TDM and IP traffic. This positions SIPxchange to reliably and cost-effectively support its rapidly expanding DID origination, Toll-Free and VoIP Termination network that provides ubiquitous Local Access Transport Area coverage in the key markets of Texas.

In need of a reliable solution that would support its existing customer base while allowing for growth throughout the United States, SIPxchange chose Dialogic for its strong leadership position in class 4 switching, support for SS7, its carrier-grade Dialogic® I-Gate® 4000 PRO Media Gateway, and an extensive array of Codecs and Protocols, including SIP, MGCP, H.323, T.38, G.711, G.729, G.723, GSM. In addition, as VoIP becomes the emerging choice for business among carriers within the United States, SIPxchange needed a solution that would allow for cost-effective direct interconnection to the Legacy PSTN Tandems. Dialogic's ControlSwitch System, with its distributed media gateway architecture, provides reliability and scalability for interconnection to the PSTN.

Grandstream Networks introduces the GXP2124 Enterprise HD IP Telephone for enterprise customers looking for a high performance HD telephone with numerous programmable keys and Electronic Hook Switch support for high call volume applications such as call centers, customer support and reception areas.

The GXP2124 is Grandstream’s first HD IP telephone with EHS support for Plantronics headsets. Users can answer and end calls using only the button on the headset – eliminating the need to touch the desktop phone. In addition, the GXP2124 features 4 line keys with up to 4 SIP accounts, 24+4 programmable speed-dial/BLF keys, broad interoperability with major SIP platforms such as Broadsoft/Asterisk/etc, superior HD audio, and 5-way conference. Grandstream is showcasing the GXP2124 and the entire family of award-winning GXP Enterprise IP Telephones at Stand B76, Hall 13 at CeBIT being held this week in Hanover, Germany.

Revolabs introduces the Revolabs FLX VoIP, the first wireless conference phone designed for VoIP networks. Supporting a wide variety of IP switches, the FLX VoIP is the only conference phone that supports the audio clarity of HD audio while providing the freedom of wireless microphones and speakers. The feature set that has been available through the Revolabs FLX for analog phone lines is now also available for IP telephone networks, providing unprecedented conference call clarity and flexibility.

The FLX VoIP integrates directly with most IP telephone switches following the SIP standard. Through this integration, new features only available through digital switch environments, such as voice mail alerts and "do not disturb," can now be offered with the FLX VoIP. The phone's wireless capabilities allow it to be used in small and midsize conference rooms without running any cables. As with the FLX for analog phone lines, this allows for a clean look while requiring less space on the conference table. The independent microphones, speaker, and dialer of the FLX VoIP give the user freedom and flexibility that other conference phone systems cannot offer.

Acrobits released its new combined Voice and Video application with SmoothFlow video technology for Apple’s iPhone.

The Voice and Video applications are available in two versions; one for the consumer market (SIP Phone) and one in a fully featured business phone application (Groundwire) with both apps loaded with features such as HD Sound, Phone book integration and Avatar Dialing. Consumers can choose from hundreds of pre-configured VoIP carriers across the world while the Groundwire business application works with both closed and open source office IP-PBX platforms such as Cisco, Avaya and Asterisk.

As the provider of the global standard in SIP training and certification, The SIP School has taught thousands of employees in the telecommunications industry how to better support their clients, products, and services. Until recently, students training to become an SIP School Certified Associate were instructed in their first session to create a SIP address with any free service. Today, The SIP School announces another option by working with OnSIP as their SIP service provider – leveraging the OnSIP API to provision each student with a SIP address on thesipschool.com domain.

OnSIP originally began SIP domain hosting to encourage their customers to simplify communications and boost their corporate branding by creating SIP addresses for employees that match their email addresses. With the OnSIP API, customers can integrate SIP address provisioning into their own service offerings as The SIP School has accomplished.

Telchemy announces DVQattest Version 2.0, a powerful new active test system for unified communications. DVQattest generates Voice over IP and Videoconferencing calls with SIP signaling, synthetic HTTP, POP3, SMTP, DNS and DHCP transactions and a range of IP network tests. This advanced test product supports pre-deployment testing, SLA monitoring and troubleshooting for converged networks and services.

DVQattest Agents are compact but highly featured software applications that can be installed on a range of operating systems and hardware platforms, including Linux servers and appliances, Android mobile phones and directly integrated into network equipment and CPE. Tests can be run on-demand or scheduled to run indefinitely. Agents can run multiple concurrent tests to other Agents or to IP phones, Web sites, Email sites and other network-based services. DVQattest Agents support complex networks with overlapping IP address spaces, VLANs and a range of SIP infrastructure configurations.

The SIP Forum announced today that it will host the second annual SIP Network Operators Conference, a two day educational conference focusing on the challenges and opportunities related to the deployment of SIP-based carrier services globally. SIPNOC US 2012 will be held in Herndon, VA on June 25-27, 2012 and will build on the success of the inaugural event last spring, which attracted leading technical and operations personnel from the global carrier community and earned high praise from attendees for its educational, non-commercial and technical content that focused on the real-world challenges operators face when deploying SIP services in global IP networks.

The SIP Forum will release further details about SIPNOC US 2012 including sign-up for early bird registration in mid-November at its conference website http://www.sipnoc.org.

snom introduces a new line of business VoIP phones – the snom 7xx series designed for both small and mid-sized businesses requiring an enterprise-class desktop phone on an SMB budget. The snom 720 and snom 760 business phones bring together the multiple programmable buttons and popular standard business functionality of the snom 3xx series with the advanced functionality, sleek styling and Gigabit Ethernet switch found in the snom 8xx series to create an advanced desktop phone at a value-driven price.

Panasonic announces that CoreDial has certified Panasonic's line of SIP Phones, the KX-TGP500, KX-TGP550, KX-UT113-B, KX-UT123-B, KX-UT133-B and KX-UT136-B, for use with their Hosted PBX telephony platform. This alliance leverages Panasonic's global leadership in the DECT cordless telephone market and CoreDial's leading private label VoIP cloud software platform, adding up to a winning combination.

Ideal for both home office and business environments, Panasonic's SIP phone systems offer the flexibility of cordless or corded models while supporting a wide range of business class features provided by the CoreDial platform. The KX-TGP500/550 systems offer convenient, cordless designs that eliminate the need to run dedicated network wiring to each employee work station while incorporating DECT 6.0 to ensure no interference with wireless networks. The new KX-UT Series is designed to complement a company's existing communication infrastructure and offer end-user savings with features including two data ports, PoE and lower power consumption while in ECO mode. All Panasonic SIP models are HD Voice enabled, allowing for outstanding voice quality, and offer flexible system expandability.

Panasonic is showcasing an extensive range of IP telephony solutions at AstriCon in Panasonic booth #204 at the Westin Westminster in Denver, Colorado, October 25-27.

As an AstriCon 2011 Gold sponsor, Panasonic is demonstrating a broad range of SIP telephones including SIP Cordless Phone Systems and KX-UT series which are interoperable with Asterisk's open source PBX. Panasonic's SIP telephones work with the Asterisk platform which offers both classical PBX functionality and advanced UC features.

In its seventh year, AstriCon is the longest running conference devoted to the Digium Asterisk communications platform. AstriCon brings together open source enthusiasts, from coders and system integrators to service providers and enterprise IT professionals, who are looking for an in-depth understanding of Asterisk open source technology.

4PSA announces the public availability of Cloud Telephony, the flexible, next-generation SIP trunking service that can be provisioned within minutes.

The new service delivers access to the telephony network using the SIP protocol and features unlimited concurrent incoming and outgoing calls. Moreover, it is possible to choose local phone numbers in over 30 countries around the world with best rates for domestic and international calls. Using VoIP technologies and the Cloud Telephony service, it is not necessary to install physical phone lines and incoming calls are always free.

new service is part of 4PSA's strategy to bring the cloud flexibility to resources that traditionally required complicated provisioning processes. Cloud Telephony follows cloud Unified Communications and cloud software licensing as a way to simplify the deployment of traditional telephony services.

Network Equipment Technologies announced that its UX Series with Session Border Controller, a qualified gateway for Microsoft Lync Server 2010, is now certified to work with AAPT's SIP trunking service. AAPT is 100% owned by Telecom New Zealand. The company's SIP voice service is a SIP trunking solution, allowing customers with an IP-enabled PBX or SIP gateway device to connect to AAPT via Ethernet and have their telephony traffic carried via IP utilizing SIP, providing a far more scaleable alternative to traditional ISDN.

Trisys introduces Replay SIP, a scalable module of its popular Replay Call Recording solution that is easily added to IP-based telephony systems. As business adds IP phones to existing telephony systems in order to take advantage of cost efficiencies, access to phone application software is often lost or requires expensive upgrade. Now with Replay SIP business can freely add call recording functionality for less than $300 per phone. The small footprint, scalable product also moves Replay to the forefront of options for new, predominantly IP phone system sales.