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| author | David Vossel <dvossel@digium.com> | 2010-08-13 20:12:22 +0000 |
|---|---|---|
| committer | David Vossel <dvossel@digium.com> | 2010-08-13 20:12:22 +0000 |
| commit | f2d6d63da2c98d75c16823eca0dce6bdef40a06c (patch) | |
| tree | cb6e9e9ae0f66ca8a2baee5be38aabafd4341862 /CHANGES | |
| parent | 0f8eaa62998c697ee72c6c3c98bf0c9478de5fb7 (diff) | |
Merged revisions 282271 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
res_stun_monitor and corresponding options CHANGES documentation
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 7 |
1 files changed, 7 insertions, 0 deletions
@@ -81,6 +81,8 @@ 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. IAX2 Changes ----------- @@ -90,6 +92,9 @@ 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 ------------ @@ -545,6 +550,8 @@ Miscellaneous * 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 ----------- |