Asterisk VoIP News

Tuesday, February 21, 2006

Dev: Test my test-branch!

The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback.

For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work.

To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down.

** PLEASE help the community, please test this branch.

Check it out like this:

SVN Checkout

Then cd into test-trunk and run "make" then "make install"

Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier.

Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information.

Thank you for your help!

P.S: Obviously, this is test code, not recommended to be closer than 2 miles from your production servers.

----- README.test-this-branch

Asterisk is developed by the user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems.

This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch.

What's in this branch?
This branch includes the following branches:

- sipdiversion: Additional support for the Diversion: header
- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
- videosupport: Improved support for video (#5427)
- peermatch: New peer matching algorithm (no bug report yet)
- rtcp: Improved support for RTCP (#2863)
- dialplan-ami-events: Report dialplan reload in manager (#5741)
- sipregister: A new registration architecture (#5834)
- subscribemwi: Support for SIP subscription of MWI notification (#6390)

Coming Here Soon:
- iptos: New IPtos support, separate audio and signalling (#6355)
- metermaids: Subscription support for parking lots (#5779)
- multiparking: Multiple parking lots (#6113)

And the following stand-alone patches
- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function

All of these exist in the bug tracker

* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
In this code, we will match incoming calls like this:

- First user on From: user name
- Then peer on From: user name *** NEW ****
- Then peer on IP and port

This means that in most configurations, you can configure a phone entry as "type=peer" instead of "type=friend". Subscriptions will work much better with just one object to match.