[asterisk-dev] Naming of the new SIP Stack

[bala murugan]

Hi ,

   I want to download the source for asterisk 12 , i am planning to
contribute to the source . can you please assist or point me the location
from where i can get the source and start working on it .

thanks,
bala


On Tue, May 21, 2013 at 1:47 PM, Matthew Jordan <mjordan at digium.com> wrote:

> Hey everyone!
>
> As you may recall, we had a discussion on this mailing list regarding
> the name of the new SIP stack [1]. If I've combed through the mailing
> list archive correctly, then we currently have the following totals for
> the name of the channel driver (using the dial string as an identifier):
>
> SIPNG: 1
> GULP: 4
> PJSIP: 9
> SIP2: 1
> SIP: 1 (with a rename of the old)
>
> It's pretty clear that at this point, barring a *lot* of additional
> commentary, PJSIP will win out. In the interest of keeping everything
> moving forward, I think it's time we call it - and start referring to
> the new SIP stack as "chan_pjsip".
>
> First, thanks to everyone who participated in the discussion!
>
> Second, if anyone would like to contribute some work to help us rename
> all of this, that would be hugely appreciated. :-)
>
> It isn't just chan_gulp that needs renaming/attention, but the SIP stack
> in the res directory as well. Functions, CLI commands, and other items
> need to be refactored. None of this is terribly hard, but it is
> certainly time consuming, and some assistance would be greatly appreciated.
>
> Thanks again!
>
> Matt
>
> [1] http://lists.digium.com/pipermail/asterisk-dev/2013-April/059669.html
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>
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