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Pingtel Announces Version 3.10 of SIPxchange

Pingtel has announced version 3.10 of its SIPxchange ECS open source VoIP solution. This new version meets the rapidly growing demand for open source IP PBX solutions as a cost-effective and flexible alternative to proprietary telephony solutions, and far exceeds the capabilities of first generation Asterisk-based open source solutions.

Adding more than 170 new high-value features, SIPxchange 3.10 advances the company's focus on meeting both the product and support needs of enterprise VoIP customers. Extending the already robust capabilities of SIPxchange, version 3.10 delivers new benchmarks with advanced telephony system features, ease of deployment and use, and scalability.

Sales of SIPxchange continue to build momentum as the market acceptance of open source IP PBX solutions grows, filling a critical void in the market for enterprise-scale IP based unified communications solutions.

Customers and independent product reviewers continue to rank SIPxchange above competing solutions for ease of use. SIPxchange installs in just hours and puts the user in control, significantly reducing deployment and administration costs, while enhancing employee productivity. Key new and enhanced usability features standard in version 3.10 include:
- An extended user control portal - Promotes individual user self-management by putting the individual user in control of key features such as time-based find-me/follow-me, voicemail management, and personal phone book, speed dial and preference management;
- New auto-attendant features – Includes a user configurable personal auto-attendant for every user on the system;
- Enhanced dial plan features – Consists of time-based dialing rules with different admin defined schedules;
- Extended presence services – Includes visual state indication and phone display;
- New group paging services – Incorporates an integrated group paging server and support for regular SIP phones and in-ceiling devices.

An intuitive Web management interface provides easy access to even the most complex features, simplifying new call center set-up, personal auto-attendant configuration and device management. Version 3.10 extends SIPxchange management capabilities to include:
- Integration with Microsoft Exchange 2007 for voicemail and Active Directory;
- Enhanced user management including user creation, phone provisioning and line assignment in only 3 clicks;
- Enhanced system administration features such as server statistics and login history reporting.

SIPxchange 3.10 fills a critical void in the market for IP-based unified communications solutions, providing an easy-to-use solution for mid-market customers. Supporting convenient VoIP system migration with a solution that scales from a single office to thousands of users, version 3.10 offers:
- Support for an unlimited number of trunk lines and up to 10,000 users per dual-server high availability system;
- Full load-sharing redundancy that maintains call connectivity even in the case of server failure;
- Automatic trunk failover and least cost routing.

The SIPxchange solution also sets the standard in terms of the number of fully plug & play managed phones and gateways. The new release supports more than 50 different phone and gateway models from vendors including Audiocodes, LG-Nortel, Polycom and Snom.

SIPxchange 3.10 will be available March 31, 2008.



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