diff options
| author | David Vossel <dvossel@digium.com> | 2010-08-13 22:27:20 +0000 |
|---|---|---|
| committer | David Vossel <dvossel@digium.com> | 2010-08-13 22:27:20 +0000 |
| commit | eca520918118816ea0ae6c86711bbdf5e5ac5b2a (patch) | |
| tree | cd817a12681d5876b7cedb0f5f38c32fe9908563 /CHANGES | |
| parent | f2d6d63da2c98d75c16823eca0dce6bdef40a06c (diff) | |
Merged revisions 282302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 4 |
1 files changed, 0 insertions, 4 deletions
@@ -1109,10 +1109,6 @@ SIP changes option is enabled, Asterisk will watch for a CNG tone in the incoming audio for a received call. If it is detected, the channel will jump to the 'fax' extension in the dialplan. - * Improved NAT and STUN support. - chan_sip now can use port numbers in bindaddr, externip and externhost - options, as well as contact a STUN server to detect its external address - for the SIP socket. See sip.conf.sample, 'NAT' section. * The default SIP useragent= identifier now includes the Asterisk version * A new option, match_auth_username in sip.conf changes the matching of incoming requests. If set, and the incoming request carries authentication info, |