Bernard.Merindol at free.fr
Thu Feb 3 06:15:25 CST 2011
On this release the bug with SIP REFER is not fixed.
See the issue https://issues.asterisk.org/view.php?id=18468
The work around is to configure SIP peer with no directmedia.
For me the version 1.8.3(rc2) is not good.
I have tested the trunk version, in this version with directmedia at yes Astreisk send 2 INVITE for re-INVITE (after transfert) with 2 CSEQ with out wait answer at the first INVITE . This is forbiden by RFC and my Aastra answser 500 ERROR at 2nd invite.
I hope is rapidly fixed, I need 1.8 for SIPS and SRTP.
On 3 févr. 2011, at 12:45, Vinícius Fontes wrote:
> I really, really hate to do this. I'm sorry in advance, but...
> I checked the roadmap for Asterisk 1.8.3 and it seems all bugs scheduled to be fixed in that release were indeed fixed. In my understanding, unless another major bug is found, RC2 will be renamed 1.8.3.
> I was unable to use the 1.8 branch in production due to the SIP REFER deadlock bug. Been using 1.8.3-rc2 in a test enviroment since its release and didn't found any issues with it yet. However I won't be allowed to upgrade our production server to 1.8.3 while it's still a RC.
> So, is there an ETA on the official release of 1.8.3? If this information is available somewhere so I don't have to write silly emails like this, I would appreciate to be pointed in the right direction.
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-dev