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.
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Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 348517 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348518 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 1f7b549..c6b7e1e 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -811,8 +811,8 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; 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.
 ;
@@ -829,7 +829,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; 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