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Note that the TCP and TLS support is currently considered experimental and

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is subject to change while we work out the remaining issues.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-03-21 15:24:43 +00:00
parent e3ea6829bb
commit a567b41083
2 changed files with 8 additions and 2 deletions
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4
CHANGES
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@@ -179,8 +179,8 @@ SIP changes
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
* Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
more information on how it is used.
* Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
configs/sip.conf.sample for more information on how it is used.
* Added a new configuration option "authfailureevents" that enables manager events when
a peer can't authenticate properly.
* Added DNS manager support to registrations not referencing a peer entry.

6
configs/sip.conf.sample
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@@ -80,6 +80,12 @@ bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
tcpenable=yes ; Enable server for incoming TCP connections (default is yes)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)