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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
1.6.0
Russell Bryant 18 years ago
parent 30c771a9d7
commit de1bcbc423
1 changed files with 4 additions and 7 deletions
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@ -42,13 +42,10 @@ CLI Changes
SIP changes SIP changes
----------- -----------
* Improved NAT and STUN support. * Improved NAT and STUN support.
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,

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