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Update documentation references in CHANGES to reflect the correct pages on the wiki.

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The current CHANGES file refers to doc/ in many places and those files no longer exist.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Sean Bright 14 years ago
parent 0fe5050553
commit 69e2a2b265
1 changed files with 33 additions and 27 deletions
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CHANGES
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@ -700,7 +700,8 @@ Applications
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 Supplementary Services for Asterisk
* 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 @@ Security Events Framework
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 @@ Miscellaneous
* 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.
@ -1306,7 +1309,8 @@ Device State Handling
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.
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 @@ Application Changes
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 @@ Miscellaneous
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 @@ Miscellaneous
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 @@ AMI - The manager (TCP/TLS/HTTP)
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 @@ SIP changes
* 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 @@ DAHDI channel driver (chan_dahdi) Changes
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 @@ Miscellaneous
* 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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