Sept 11, 2017

sipXcom open source support

Summary

eZuce is pleased to announce the General Availability of sipXcom 17.08.

We’ve been busy since 17.04! While we’ve been working hard on the next generation ‘Docker-ized’ version sipXcom (see 17.08.docker branch) we’ve added quite a large collection of new features and enhancements for the 17.08 release. These new features include SIP Proxy congestion management tools and configuration support for some new phones. The new user portal continues to get additional enhancements including the ability for users to upload their greetings as MP3 or WAV files.

Big thanks to IANT for a raft of Polycom and Yealink updates and to João Veríssimo for his work on the Grandstream plugins. An eZuce customer helped to fund and contribute back additional work for Grandstream and Zoiper phone plugins.

Also as always, the Dev & QA teams at eZuce have done excellent work on this release!

In all 45 issues (enhancements / fixes) are addressed for sipXcom in this beta release.

In a break from our releases every 4 months, we’re planning on a 17.10 release as our next release. This will be a release focused on a new option for anchoring calls in FreeSWITCH.

Highlights

sipXcom New Features:

  • Zoiper Provisioning Support
  • G.729 Codec included
  • Optional Retry-After header in Proxy 503 Responses when Overloaded
  • Proxy Congestion Management feature
  • Grandstream 2130, 2140 and 2160 Phone Templates
  • Allow MP3 or WAV User Greeting Upload in New User Portal User Settings

sipXcom Improvements:

  • Improvements to Yealink phone configurations
    • AutoProvision service now supports Yealink phones
    • Support for Cisco Discovery Protocol settings
    • Phone power savings setting for firmware 8.x and later.
    • Local DTMF Tone parameter
  • Improvements to Polycom phone configurations
    • RealPresence Trio Firmware Support
    • Call Waiting Behavior
    • Device Base Profile (Ensure Generic for phones that may have shipped as Lync phones)
    • Firmware 5.5.2 support
  • Display calling number and caller-id in UniteWeb Voicemail page
  • Flexible automatic Phone Line label generation for Polycom phones
  • Allow SipRedirectorPickUp port to bind to other TCP/IP Ports
  • gridfs-voicemail-cli.jar command line tool now allows upload of voicemail greeting
  • New REST Calls

Notes

  1. Full Release Notes with installation information are located here: http://wiki.sipxcom.org/display/sipXcom/sipXcom+17.08

Who Should Install?

This release is recommended for all 4.6 and later installations.

Questions

Please post to the sipXcom-users google group if you have questions.

https://groups.google.com/forum/#!forum/sipxcom-users

Specific Issues Addressed

Specific issues can be located in the detailed release notes in the wiki at: http://wiki.sipxcom.org/display/sipXcom/sipXcom+17.08