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remove a couple of entries that got duplicated and snuck into the SIP section. Also, align the NAT/STUN entry with the others.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2007-07-25 01:06:02 +00:00
parent 30c771a9d7
commit de1bcbc423
1 changed files with 4 additions and 7 deletions
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@@ -46,9 +46,6 @@ SIP changes
chan_sip now can use port numbers in bindaddr, externip and externhost 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 options, as well as contact a STUN server to detect its external address
for the SIP socket. See sip.conf.sample, 'NAT' section. for the SIP socket. See sip.conf.sample, 'NAT' section.
* Added the bindaddr option to gtalk.conf.
* Added the ability to specify arguments to the Dial application when using
the DUNDi switch in the dialplan.
* The default SIP useragent= identifier now includes the Asterisk version * 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. * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
If set, and the incoming request carries authentication info, If set, and the incoming request carries authentication info,