[asterisk-dev] 1.8.3?

Leif Madsen leif.madsen at asteriskdocs.org
Thu Feb 3 10:28:32 CST 2011

On 11-02-03 07:15 AM, Bernard Merindol(F) wrote:
> 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.

I will be reviewing the issues mentioned here, but if they are not regressions 
from the previous release, then we'll need to move on and get those resolved in 
a future sprint cycle.


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