[asterisk-dev] SIPit18 tests - no severe damage discovered!

[Olle E Johansson]

Friends, testers,

This week has been totally dedicated to testing. I've learned so many  
testing methods, so I can keep
you fully occupied for weeks and months. Just to be nice, before I  
start the new test programme, I will
let you rest this weekend. In tribute to Japan, eat some sushi, drink  
sake and relax...

Last fall, I brought Asterisk to SIPit for the first time. I  
survived, Asterisk survived - but I had a long
list of things to fix. Some very serious errors, some smaller syntax  
errors and some minor things.

In 30 minutes, we're closing SIPit 18 in Tokyo. This week makes me  
very proud over the Asterisk
development of the version 1.4 SIP stack. The errors I've found are  
all very minor and easy to fix.
I have some larger issues that need to be resolved, but those does  
not affect normal communication
unless you have some seriously advanced networks. I even found errors  
in other product's SIP stacks!!!

So the Asterisk SIP team is making progress, proven by this week's  
tests. And now the SIP community
here expect us to do even better next time - adding new SIP features  
and functions.  New stuff for you
to test!

Due to all the changes I made in the SIP code this week, mostly  
integration functions from the SIPtransfer
branch, and the changes made in other modules of Asterisk the test  
branch is now temporarily out of
synch with svn trunk. I will do my best to restore it early next  
week, if not earlier.

As soon as I've finished integrating the SIP transfer code, the SIP  
channel will go into bug fixing mode
for the 1.4 release - only adding functions that exist in the bug  
tracker (pending review) and has been
tested. At that time, I'll fork and create a parallell development  
branch for chan_sip3. I will start that
branch with integrating the sipregister and sippeers branches, as  
well as some other new code that
I have in other branches. After that, we'll look into transactions,  
transport layer awareness (UDP/TCP)
and adding new features. More about chan_sip3 later.

Let me finish with a big thank you to all the people that have  
supported the work with the SIP stack
- all bug reporters, sponsors, all testers, coders and users. We are  
moving forward together.
Asterisk 1.4 will be a great product!

Greetings from Tokyo!
/Olle

PS. Well, the test branch still compiles. So as an afterthought -  
keep testing! We need all
       bug reports, patches, fixes. Read the README.test-this-branch  
and test, test, test!
       You're not on hold :-)