[asterisk-dev] Asterisk Supertest in Tokyo!
Masakazu Nakano
n-mack at md.neweb.ne.jp
Sun Apr 16 20:49:27 MST 2006
Hi OEJ
I was developed japanese SIP provider such as FusionCommunications
SIP stack and that is already approval by them.
In Japan,there is a couple of SIP implementation is avaiable.
Usually their SIP isn't 100% pure RFC3261.
I know some issue about session-timer and RFC3325 in NTT's one.
and I think bit difficult to implement to svn.
because they aren't agree to disclose their additional SIP
implementation in NDA.
BTW,Do you have a time to meet with me? :-)
WBR
Masakazu Nakano.
http://www.dairiten.com/
mack at dairiten.com
On Mon, 17 Apr 2006 04:15:13 +0200
Olle E Johansson <oej at edvina.net> wrote:
> Ok testers,
>
> So I forgot about y'all again for another weekend, leaving you stuck
> with family life or alone with your TV-set
> and 247.6 channels.
>
> My apologies.
>
> The reason? Another Asterisk-related business trip, this time to the
> far east! I am in Tokyo, Japan, testing
> Asterisk at SIPit18 - the international SIP interoperability test
> event organized by the SIP forum.
>
> This is a huge test lab - imagine a large conference hall filled with
> almost 200 people, each one with
> laptops - one or several - phones, video cameras, hubs, coke bottles
> and other pieces of important stuff that I
> am not allowed to tell y'all about (sorry for the Huntsville accent
> there)...
>
> I will spend one week testing Asterisk with many, many SIP devices to
> make sure we pinpoint problems and
> hopefully fix them too. There are teams here with impressive test
> equipment that test almost every possible
> construction in the SIP protocol and addons. Stressing and fun!
>
> I hope to merge the final parts of the SIP transfer branch into svn
> trunk today, so that it exists both in trunk
> and the test branch. During this week, you will propably see some
> other bug fixes being integrated into
> the svn repositories, if needed both in 1.2 and trunk, in some cases
> only in trunk.
>
> This week of testing will lead to an even better Asterisk SIP stack!
> My participation is generously sponsored by
> Digium and Voop - Thank You!
>
> /Olle
>
> PS. This does not mean that you're off the hook. Keep testing!
> A special thank you to Max Bressel for test reports for the
> "sipregister" part of the test branch!!!
> Max - you're the Asterisk Tester of the week!
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list