Looks like it is part of the 1.6 Beta.
>From the Change Log:
2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <[EMAIL PROTECTED]>
* CREDITS, include/asterisk/http.h, main/tcptls.c (added),
main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
configs/sip.conf.sample, CHANGES: Merge changes from
team/group/sip-tcptls This set of changes introduces TCP and TLS
support for chan_sip. There are various new options in
configs/sip.conf.sample that are used to enable these features.
Also, there is a document, doc/siptls.txt that describes some
things in more detail. This code was implemented by Brett Bryant
and James Golovich. It was reviewed by Joshua Colp and myself. A
number of other people participated in the testing of this code,
but since it was done outside of the bug tracker, I do not have
their names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what
exists there.)
On Feb 13, 2008 4:21 PM, Razza <[EMAIL PROTECTED]> wrote:
> I am aware there is a SIP over TCP patch. Will this ever become part of
> a release, if so are there any timelines?
> Thanks in advance.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
