[asterisk-commits] wdoekes: branch 11 r396995 - in /branches/11: ./ configs/

[SVN commits to the Asterisk project]

Author: wdoekes
Date: Tue Aug 20 06:47:16 2013
New Revision: 396995

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=396995
Log:
Add "autoframing" option to sip.conf.sample and h323.conf.sample.

The autoframing option was added to chan_sip.c in r43243 (mogorman,
2006-09-19 01:32:57), but never made its way into the sample configs.

Review: https://reviewboard.asterisk.org/r/2768/
........

Merged revisions 396994 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Modified:
    branches/11/   (props changed)
    branches/11/configs/h323.conf.sample
    branches/11/configs/sip.conf.sample

Propchange: branches/11/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: branches/11/configs/h323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/11/configs/h323.conf.sample?view=diff&rev=396995&r1=396994&r2=396995
==============================================================================
--- branches/11/configs/h323.conf.sample (original)
+++ branches/11/configs/h323.conf.sample Tue Aug 20 06:47:16 2013
@@ -29,6 +29,8 @@
 ;allow=gsm		; Always allow GSM, it's cool :)
 ;allow=ulaw		; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 			; for framing options
+;autoframing=yes	; Set packetization based on the remote endpoint's (ptime)
+			; preferences. Defaults to no.
 ;
 ; User-Input Mode (DTMF)
 ;

Modified: branches/11/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=diff&rev=396995&r1=396994&r2=396995
==============================================================================
--- branches/11/configs/sip.conf.sample (original)
+++ branches/11/configs/sip.conf.sample Tue Aug 20 06:47:16 2013
@@ -310,6 +310,8 @@
 ;allow=ulaw                     ; Allow codecs in order of preference
 ;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 				; for framing options
+;autoframing=yes		; Set packetization based on the remote endpoint's (ptime)
+				; preferences. Defaults to no.
 ;
 ; This option specifies a preference for which music on hold class this channel
 ; should listen to when put on hold if the music class has not been set on the
@@ -1192,6 +1194,7 @@
 ; language
 ; allow
 ; disallow
+; autoframing
 ; insecure
 ; trustrpid
 ; progressinband