diff options
author | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-09 18:51:52 +0000 |
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committer | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-09 18:51:52 +0000 |
commit | 722eb3c4c3cfa1c0cee915c949c5f95199ee24dd (patch) | |
tree | 25683963c5e51bdedd6211cd0ea92a85639505c3 /CHANGES | |
parent | 815b5b09da5e555add7bba3d8fca588e7611248a (diff) |
Merged revisions 285710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 17 |
1 files changed, 13 insertions, 4 deletions
@@ -73,6 +73,11 @@ SIP Changes RTP has been outfitted with the same abilities. * Added support for setting the Max-Forwards: header in SIP requests. Setting is available in device configurations as well as in the dial plan. + * Addition of the 'subscribe_network_change' option for turning on and off + res_stun_monitor module support in chan_sip. + * Addition of the 'auth_options_requests' option for turning on and off + authentication for OPTIONS requests in chan_sip. + IAX2 Changes ----------- @@ -82,6 +87,9 @@ IAX2 Changes encryption is being used. This interoperates with the SIP SRTP implementation so that a secure SIP call can be bridged to a secure IAX call when the dialplan requires bridged channels to be "secure". + * Addition of the 'subscribe_network_change' option for turning on and off + res_stun_monitor module support in chan_iax. + MGCP Changes ------------ @@ -535,6 +543,10 @@ Miscellaneous * The UNISTIM channel driver (chan_unistim) has been updated to support devices that have less than 3 lines on the LCD. * Realtime now supports database failover. See the sample extconfig.conf for details. + * The addition of improved translation path building for wideband codecs. Sample + rate changes during translation are now avoided unless absolutely necessary. + * The addition of the res_stun_monitor module for monitoring and reacting to network + changes while behind a NAT. CLI Changes ----------- @@ -550,6 +562,7 @@ CLI Changes manager.conf. * Added 'all' keyword to the CLI command "channel request hangup" so that you can send the channel hangup request to all channels. + * Added a "core reload" CLI command that executes a global reload of Asterisk. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 ------------- @@ -1091,10 +1104,6 @@ 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, |