diff options
author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-13 23:43:06 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-13 23:43:06 +0000 |
commit | 0c6b474f1554c67e390956f382a7c67e8924507e (patch) | |
tree | cdd6ec020879aeb6834c250c58d7bc93a80dadb0 /CHANGES | |
parent | 02a222b4cc4b2eabee2193ea1ee39162c433f218 (diff) |
- Break up the Misc. section a bit with a new section for Misc. New Modules
- Change spacing a bit in some places for consistent indentation
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98656 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 140 |
1 files changed, 72 insertions, 68 deletions
@@ -43,22 +43,22 @@ AMI - The manager (TCP/TLS/HTTP) Dialplan functions ------------------ * Added the DEVICE_STATE() dialplan function which allows retrieving any device - state in the dialplan, as well as creating custom device states that are - controllable from the dialplan. + state in the dialplan, as well as creating custom device states that are + controllable from the dialplan. * Extend CALLERID() function with "pres" and "ton" parameters to fetch string representation of calling number presentation indicator and numeric representation of type of calling number value. * MailboxExists converted to dialplan function * A new option to Dial() for telling IP phones not to count the call - as "missed" when dial times out and cancels. + as "missed" when dial times out and cancels. * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan - mutex. No deadlocks are possible, as LOCK() only allows a single lock to be - held for any given channel. Also, locks are automatically freed when a - channel is hung up. + mutex. No deadlocks are possible, as LOCK() only allows a single lock to be + held for any given channel. Also, locks are automatically freed when a + channel is hung up. * Added HINT() dialplan function that allows retrieving hint information. - Hints are mappings between extensions and devices for the sake of - determining the state of an extension. This function can retrieve the list - of devices or the name associated with a hint. + Hints are mappings between extensions and devices for the sake of + determining the state of an extension. This function can retrieve the list + of devices or the name associated with a hint. * Added EXTENSION_STATE() dialplan function which allows retrieving the state of any extension. * Added SYSINFO() dialplan function which allows retrieval of system information @@ -114,13 +114,13 @@ SIP changes states it is not needed. For phones, however, that do require it the "registertrying" option has been added so it can be enabled. * A new option called "callcounter" (global/peer/user level) enables call counters needed - for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously - used to enable this functionality). + for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously + used to enable this functionality). * New settings for timer T1 and timer B on a global level or per device. This makes it - possible to force timeout faster on non-responsive SIP servers. These settings are - considered advanced, so don't use them unless you have a problem. + possible to force timeout faster on non-responsive SIP servers. These settings are + considered advanced, so don't use them unless you have a problem. * Added a dial string option to be able to set the To: header in an INVITE to any - SIP uri. + SIP uri. * Added a new global and per-peer option, qualifyfreq, which allows you to configure the qualify frequency. @@ -150,11 +150,6 @@ Console Channel Driver changes * Added experimental support for video send & receive to chan_oss. This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as a video source. - * 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 - interfaces, as well. Note that this channel driver requires v19 or higher - of portaudio; older versions have a different API. Phone channel changes (chan_phone) ---------------------------------- @@ -181,8 +176,8 @@ Zaptel channel driver (chan_zap) Changes ---------------------------------------- * SS7 support in chan_zap (via libss7 library) * In India, some carriers transmit CID via dtmf. Some code has been added - that will handle some situations. The cidstart=polarity_IN choice has been added for - those carriers that transmit CID via dtmf after a polarity change. + that will handle some situations. The cidstart=polarity_IN choice has been added for + those carriers that transmit CID via dtmf after a polarity change. * CID matching information is now shown when doing 'dialplan show'. * Added zap show version CLI command to chan_zap. * Added setvar support to zapata.conf channel entries. @@ -192,21 +187,26 @@ Zaptel channel driver (chan_zap) Changes event indicating the new state of the mailbox is also generated, so that the normal MWI facilities in Asterisk work as usual. * Added signalling type 'auto', which attempts to use the same signalling type - for a channel as configured in Zaptel. This is primarily designed for analog - ports, but will also work for digital ports that are configured for FXS or FXO - signalling types. This mode is also the default now, so if your zapata.conf - does not specify signalling for a channel (which is unlikely as the sample - configuration file has always recommended specifying it for every channel) then - the 'auto' mode will be used for that channel if possible. + for a channel as configured in Zaptel. This is primarily designed for analog + ports, but will also work for digital ports that are configured for FXS or FXO + signalling types. This mode is also the default now, so if your zapata.conf + does not specify signalling for a channel (which is unlikely as the sample + configuration file has always recommended specifying it for every channel) then + the 'auto' mode will be used for that channel if possible. * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb - state for a channel; also ensured that the DNDState Manager event is - emitted no matter how the DND state is set or cleared. + state for a channel; also ensured that the DNDState Manager event is + emitted no matter how the DND state is set or cleared. -A new channel driver: Unistim ------------------------------ +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_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 + interfaces, as well. Note that this channel driver requires v19 or higher + of portaudio; older versions have a different API. DUNDi changes ------------- @@ -350,8 +350,8 @@ Music On Hold Changes to this music on hold class. * Support for realtime music on hold has been added. * In conjunction with the realtime music on hold, a general section has - been added to musiconhold.conf, its sole variable is cachertclasses. If this - is set, then music on hold classes found in realtime will be cached in memory. + been added to musiconhold.conf, its sole variable is cachertclasses. If this + is set, then music on hold classes found in realtime will be cached in memory. AEL Changes ----------- @@ -373,11 +373,11 @@ AEL Changes fashion: Set(LOCAL(myvar)=someval); ("local" is now an AEL keyword). * utils/conf2ael introduced. Will convert an extensions.conf - file into extensions.ael. Very crude and unfinished, but - will be improved as time goes by. Should be useful for a - first pass at conversion. + file into extensions.ael. Very crude and unfinished, but + will be improved as time goes by. Should be useful for a + first pass at conversion. * aelparse will now read extensions.conf to see if a referenced - macro or context is there before issueing a warning. + macro or context is there before issueing a warning. Call Features (res_features) Changes ------------------------------------ @@ -417,39 +417,10 @@ Logger changes and to ensure that the oldest log file gets deleted. * Added realtime support for the queue log -Miscellaneous -------------- - * Ability to use libcap to set high ToS bits when non-root - on Linux. If configure is unable to find libcap then you - can use --with-cap to specify the path. - * Added maxfiles option to options section of asterisk.conf which allows you to specify - what Asterisk should set as the maximum number of open files when it loads. - * Added the jittertargetextra configuration option. +Miscellaneous New Modules +------------------------- * Added a new CDR module, cdr_sqlite3_custom. - * The cdr_manager module has a [mappings] feature, like cdr_custom, - to add fields to the manager event from the CDR variables. * Added a new realtime configuration module, res_config_sqlite - * 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. - * 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. - It allows you to configure a prefix for auto-monitor recordings. - * Added support for writing and running your dialplan in lua. See - configs/extensions.lua.sample for examples of how to do this. - * A new extension pattern matching algorithm, based on a trie, is introduced - here, that could noticeably speed up mid-sized to large dialplans. - It is NOT used by default, as duplicating the behaviour of the old pattern - matcher is still under development. A config file option, in extensions.conf, - in the [general] section, called "extenpatternmatchingnew", is by default - set to false; setting that to true will force the use of the new algorithm. - Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can - be used to switch the algorithms at run time. - * A new option when starting a remote asterisk (rasterisk, asterisk -r) for - specifying which socket to use to connect to the running Asterisk daemon - (-s) * Added a new codec translation module, codec_resample, which re-samples signed linear audio between 8 kHz and 16 kHz to help support wideband codecs. @@ -473,3 +444,36 @@ Miscellaneous on as the channel's audio. This is very useful for building custom vocoders or doing recording or analysis of the channel's audio in another application. + +Miscellaneous +------------- + * Ability to use libcap to set high ToS bits when non-root + on Linux. If configure is unable to find libcap then you + can use --with-cap to specify the path. + * Added maxfiles option to options section of asterisk.conf which allows you to specify + what Asterisk should set as the maximum number of open files when it loads. + * Added the jittertargetextra configuration option. + * The cdr_manager module has a [mappings] feature, like cdr_custom, + to add fields to the manager event from the CDR variables. + * 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. + * 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. + It allows you to configure a prefix for auto-monitor recordings. + * Added support for writing and running your dialplan in lua. See + configs/extensions.lua.sample for examples of how to do this. + * A new extension pattern matching algorithm, based on a trie, is introduced + here, that could noticeably speed up mid-sized to large dialplans. + It is NOT used by default, as duplicating the behaviour of the old pattern + matcher is still under development. A config file option, in extensions.conf, + in the [general] section, called "extenpatternmatchingnew", is by default + set to false; setting that to true will force the use of the new algorithm. + Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can + be used to switch the algorithms at run time. + * A new option when starting a remote asterisk (rasterisk, asterisk -r) for + specifying which socket to use to connect to the running Asterisk daemon + (-s) + |