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

[trixter aka Bret McDanel]

On Wed, 2006-02-01 at 12:05 +0100, Olle E Johansson wrote:
> trixter aka Bret McDanel wrote:
> > On Wed, 2006-02-01 at 11:28 +0100, Olle E Johansson wrote:
> > 
> >>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.
> > 
> > 
> > not to demean your work but why cant an existing sip stack be used like
> > freeswitch.org is doing?  There are good stacks out there that do all
> > this, and as such it wouldnt take as much work becuase you wouldnt have
> > to reinvent the wheel.  
> > 
> That is a decision for the future, for chan_sip3. Most of this work is 
> done and targeted at current chan_sip.

maybe its the cough syrup but I dont understand what you just said.

What specifically is a decision for the future?  Using an alternate sip
stack or yours?

When you said 'most of this work is done' did you mean yours or
integrating a 3rd party already working sip stack?

I think what you are saying, and please correct me if I am wrong, is
that most of what you are doing in terms of these features is already
done, and that its to work with the current chan_sip, basically just
adding features to it.

Is that correct?

-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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