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remove current STUN support from chan_sip.c

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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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2010-08-13 22:23:38 +00:00
parent 5b3270acc2
commit 22682c2eee
4 changed files with 15 additions and 63 deletions
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@@ -1101,10 +1101,6 @@ 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,