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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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -797,8 +797,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; However, this is only useful if the external traffic can reach us.
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; The following settings are allowed (both globally and in individual sections):
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;
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; nat = no ; Default. Use rport if the remote side says to use it.
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; nat = force_rport ; Force rport to always be on.
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; nat = no ; Use rport if the remote side says to use it.
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; nat = force_rport ; Force rport to always be on. (default)
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; nat = yes ; Force rport to always be on and perform comedia RTP handling.
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; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling.
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;
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@@ -815,7 +815,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; by outside parties as Asterisk will respond to different ports for defined and
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; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
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; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
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; other, then valid users with settings differing from those in the general section will
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; other, then valid peers with settings differing from those in the general section will
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; be discoverable.
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;
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; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
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