[asterisk-dev] SIP transfer committed

[kibeki at gmx.net]

Hi,
we want to test the new SIP transfer. What do i have to checkout to get the 
necessary code? currently we are using the latest release versions (1.2.8).
much thanks and regards
Bernd

Olle E Johansson wrote:
> Friends,
> 
> I finally committed the last piece of the new SIP transfer support code. 
> This greatly enhances the support of SIP
> transfers - or at least is meaning to. The code has been tested on 1.2 
> for almost a year in production, but the
> trunk version is a port from this. A port in many cases means that one 
> introduces new bugs.
> 
> This code will be tested heavily this week. If you test it and find new 
> bugs, please make sure you report
> them on the bug tracker so we can fix this quickly.
> 
> This patch affected both the handling of incoming INVITEs and REFERs. We 
> now have support for INVITE/Replaces
> in relation to REFER. We will have to test what needs to be done to 
> support INVITE/Replaces in relation to call pickup,
> that is an area I have not been focusing on yet. As kpfleming says: I 
> demand that is may or may not work :-)
> 
> Thanks to Nuvio, Voop and Foniris for sponsoring the work with SIP 
> transfers. It's been a long journey through
> masquerading, channel locks and a lot of evil bugs.
> 
> /O
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