[asterisk-commits] seanbright: trunk r365213 - /trunk/CHANGES
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 3 13:43:58 CDT 2012
Author: seanbright
Date: Thu May 3 13:43:54 2012
New Revision: 365213
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=365213
Log:
Update documentation references in CHANGES to reflect the correct pages on the wiki.
The current CHANGES file refers to doc/ in many places and those files no longer exist.
Modified:
trunk/CHANGES
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=365213&r1=365212&r2=365213
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Thu May 3 13:43:54 2012
@@ -700,7 +700,8 @@
notices a change.
* Voicemail now includes rdnis within msgXXXX.txt file.
* ExternalIVR now supports IPv6 addresses.
- * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
+ * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
+ at https://wiki.asterisk.org/wiki/x/oQBB
* ParkedCall and Park can now specify the parking lot to use.
Dialplan Functions
@@ -1001,7 +1002,7 @@
* Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
DAHDI/ISDN supports call completion for the following switch types:
EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
- See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
+ See https://wiki.asterisk.org/wiki/x/2ABQ for details.
Multicast RTP Support
---------------------
@@ -1019,7 +1020,8 @@
sends these events to the "security" logger level. Currently, AMI is the only
Asterisk component that reports security events. However, SIP support will be
coming soon. For more information on the security events framework, see the
- "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
+ "Asterisk Security Framework" section of the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/wgBQ
* SIP support was added in Asterisk 10
* This API now supports IPv6 addresses
@@ -1072,7 +1074,8 @@
* The Realtime dialplan switch now caches entries for 1 second. This provides a
significant increase in performance (about 3X) for installations using this switchtype.
* Distributed devicestate now supports the use of the XMPP protocol, in addition to
- AIS. For more information, please see doc/distributed_devstate-XMPP.txt
+ AIS. For more information, please see the Distributed Device State section of the
+ Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
* The addition of G.719 pass-through support.
* Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
during device configuration.
@@ -1300,13 +1303,14 @@
Device State Handling
---------------------
* The event infrastructure in Asterisk got another big update to help support
- distributed events. It currently supports distributed device state and
- distributed Voicemail MWI (Message Waiting Indication). A new module has
- been merged, res_ais, which facilitates communicating events between servers.
- It uses the SAForum AIS (Service Availability Forum Application Interface
- Specification) CLM (Cluster Management) and EVT (Event) services to maintain
- a cluster of Asterisk servers, and to share events between them. For more
- information on setting this up, see doc/distributed_devstate.txt.
+ distributed events. It currently supports distributed device state and
+ distributed Voicemail MWI (Message Waiting Indication). A new module has
+ been merged, res_ais, which facilitates communicating events between servers.
+ It uses the SAForum AIS (Service Availability Forum Application Interface
+ Specification) CLM (Cluster Management) and EVT (Event) services to maintain
+ a cluster of Asterisk servers, and to share events between them. For more
+ information on setting this up, refer to the Distributed Device State section
+ of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
Dialplan Functions
------------------
@@ -1381,8 +1385,8 @@
the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
change to whisper mode, and pressing 6 will change to barge mode.
* ExternalIVR now takes several options that affect the way it performs, as
- well as having several new commands. Please see doc/externalivr.txt for the
- complete documentation.
+ well as having several new commands. Please see the External IVR page on the Asterisk
+ wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
* Added ability to communicate over a TCP socket instead of forking a child process for the
ExternalIVR application.
* ChanIsAvail has a new option, 'a', which will return all available channels instead
@@ -1501,8 +1505,9 @@
the 'setvar' option to cause a given audio file to be played upon completion
of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
Skinny channels only.
- * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
- for more information.
+ * You can now compile Asterisk against the Hoard Memory Allocator, see the
+ Hoard page on the Asterisk wiki for more information:
+ https://wiki.asterisk.org/wiki/x/pQBB
* Config file variables may now be appended to, by using the '+=' append
operator. This is most helpful when working with long SQL queries in
func_odbc.conf, as the queries no longer need to be specified on a single
@@ -1520,7 +1525,7 @@
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
* Manager has undergone a lot of changes, all of them documented
- in doc/manager_1_1.txt
+ on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
* Manager version has changed to 1.1
* Added a new action 'CoreShowChannels' to list currently defined channels
and some information about them.
@@ -1567,10 +1572,10 @@
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.
* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
- * New command: Atxfer. See doc/manager_1_1.txt for more details or
- manager show command Atxfer from the CLI
- * New command: IAXregistry. See doc/manager_1_1.txt for more details or
- manager show command IAXregistry from the CLI
+ * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
+ or manager show command Atxfer from the CLI
+ * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
+ details or manager show command IAXregistry from the CLI
Dialplan functions
------------------
@@ -1682,8 +1687,8 @@
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
- * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
- configs/sip.conf.sample for more information on how it is used.
+ * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
+ and configs/sip.conf.sample for more information on how it is used.
* Added a new configuration option "authfailureevents" that enables manager events when
a peer can't authenticate properly.
* Added DNS manager support to registrations for peers not referencing a peer entry.
@@ -1773,9 +1778,10 @@
New Channel Drivers
-------------------
- * Added a new channel driver, chan_unistim. See doc/unistim.txt and
- configs/unistim.conf.sample for details. This new channel driver allows
- you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
+ * Added a new channel driver, chan_unistim. See the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
+ for details. This new channel driver allows you to use Nortel i2002,
+ i2004, and i2050 phones with Asterisk.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
work with Mac CoreAudio, but portaudio supports a number of other audio
@@ -2147,8 +2153,8 @@
* Added the jittertargetextra configuration option.
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
- This information is also documented in doc/qos.tex, or the IP Quality of Service
- section of asterisk.pdf.
+ This information is also documented on the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/EYBG
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
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