[asterisk-biz] Sponsor Asterisk development project: SIP attended transfers

[Olle E Johansson]

Last year, I spent a lot of time rewriting SIP transfers in chan_sip, in 
order to enhance the support for attended transfers, especially in the 
case where two servers where supported. This was paid for by a service 
provider, who after they installed it in their production systems 
decided not to pay all of the fees that we had agreed upon.

Even though it works in their environment, there is still some work to 
do to finish this quite large change and make it more generic and 
complete for standalone servers, as well as cleaning the source code up 
for peer review and possible commit. I need funding up to $10.000 USD to 
be able to dedicate time to complete this project and submit it to the 
Asterisk project.

The code
* Makes transfers fail properly (right now, we send the transfer target
   a congestion tone or a busy signal, with no recovery)
* Adds a scheme to allow/deny transfers per peer/user or in general
* Adds support for INVITE with replaces headers
* Adds support for attended transfers with call legs on different servers

Most of this have been tested in depth, and is in production today.

I already have one service provider contributing 1.000 USD towards this 
project. I will start as soon as I have guarantees of 10.000 USD, not to 
operate at a loss again with this project. Any contribution is welcome.

If you are interested, please mail me off list.

Thank you for your support!

/Olle