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 parallel 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