[asterisk-commits] kpfleming: branch 10 r348517 - in /branches/10: ./ configs/sip.conf.sample

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Dec 18 12:28:24 CST 2011


Author: kpfleming
Date: Sun Dec 18 12:28:20 2011
New Revision: 348517

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=348517
Log:
Correct two flaws in sip.conf.sample related to AST-2011-013.

* The sample file listed *two* values for the 'nat' option as being the default.
  Only 'force_rport' is the default.

* The warning about having differing 'nat' settings confusingly referred to both
  peers and users.
........

Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........

Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Modified:
    branches/10/   (props changed)
    branches/10/configs/sip.conf.sample

Propchange: branches/10/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: branches/10/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/10/configs/sip.conf.sample?view=diff&rev=348517&r1=348516&r2=348517
==============================================================================
--- branches/10/configs/sip.conf.sample (original)
+++ branches/10/configs/sip.conf.sample Sun Dec 18 12:28:20 2011
@@ -808,8 +808,8 @@
 ; However, this is only useful if the external traffic can reach us.
 ; The following settings are allowed (both globally and in individual sections):
 ;
-;        nat = no                ; Default. Use rport if the remote side says to use it.
-;        nat = force_rport       ; Force rport to always be on.
+;        nat = no                ; Use rport if the remote side says to use it.
+;        nat = force_rport       ; Force rport to always be on. (default)
 ;        nat = yes               ; Force rport to always be on and perform comedia RTP handling.
 ;        nat = comedia           ; Use rport if the remote side says to use it and perform comedia RTP handling.
 ;
@@ -826,7 +826,7 @@
 ; by outside parties as Asterisk will respond to different ports for defined and
 ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
 ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
-; other, then valid users with settings differing from those in the general section will
+; other, then valid peers with settings differing from those in the general section will
 ; be discoverable.
 ;
 ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by




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