[svn-commits] lmadsen: branch 1.4 r226382 - /branches/1.4/configs/sip.conf.sample

[SVN commits to the Digium repositories]

Author: lmadsen
Date: Wed Oct 28 15:06:13 2009
New Revision: 226382

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=226382
Log:
Update documentation in sip.conf.sample.

Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.

(closes issue #15644)
Reported by: lmadsen

Modified:
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=226382&r1=226381&r2=226382
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Wed Oct 28 15:06:13 2009
@@ -360,6 +360,13 @@
                                 ; at call setup (a new feature in 1.4 - setting up the
                                 ; call directly between the endpoints instead of sending
                                 ; a re-INVITE).
+
+                                ; Additionally this option does not disable all reINVITE operations.
+                                ; It only controls Asterisk generating reINVITEs for the specific
+                                ; purpose of setting up a direct media path. If a reINVITE is
+                                ; needed to switch a media stream to inactive (when placed on
+                                ; hold) or to T.38, it will still be done, regardless of this
+                                ; setting.
 
 ;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                 ; the call directly with media peer-2-peer without re-invites.