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Merged revisions 110499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -149,8 +149,8 @@ SIP changes
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* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
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were not properly torn down due to network or endpoint failures during an established
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SIP session.
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* Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
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more information on how it is used.
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* Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
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configs/sip.conf.sample for more information on how it is used.
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IAX2 changes
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------------
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@@ -80,6 +80,12 @@ bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
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; bindport is the local UDP port that Asterisk will listen on
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bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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;
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; Note that the TCP and TLS support for chan_sip is currently considered
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; experimental. Since it is new, all of the related configuration options are
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; subject to change in any release. If they are changed, the changes will
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; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
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;
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tcpenable=yes ; Enable server for incoming TCP connections (default is yes)
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tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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