diff options
author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-04-23 15:34:51 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-04-23 15:34:51 +0000 |
commit | 61903011dd5b1ecf9576c679b0ce5fe5e59c10f5 (patch) | |
tree | 262921feb1cf638ae6682ccbe743db81c0f6f99a /CHANGES | |
parent | f0cc154be08aee879734f2461166ab2caaf3f9d2 (diff) |
Add OSP support for IAX2 to the changes file. Also, slightly reorganize some
of the content.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61760 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 145 |
1 files changed, 75 insertions, 70 deletions
@@ -2,6 +2,81 @@ --- Functionality changes since Asterisk 1.4-beta was branched ---------------- ------------------------------------------------------------------------------- +AMI - The manager (TCP/TLS/HTTP) +-------------------------------- + * Added the URI redirect option for the built-in HTTP server + * The output of CallerID in Manager events is now more consistent. + CallerIDNum is used for number and CallerIDName for name. + * enable https support for builtin web server. + See configs/http.conf.sample for details. + * Added a new action, GetConfigJSON, which can return the contents of an + Asterisk configuration file in JSON format. This is intended to help + improve the performance of AJAX applications using the manager interface + over HTTP. + * SIP and IAX manager events now use "ChannelType" in all cases where we + indicate channel driver. Previously, we used a mixture of "Channel" + and "ChannelDriver" headers. + * Added a "Bridge" action which allows you to bridge any two channels that + are currently active on the system. + +Dialplan functions +------------------ + * Added the DEVSTATE() dialplan function which allows retrieving any device + 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 + +CLI Changes +----------- + * New CLI command "core show settings" + * Added 'core show channels count' CLI command. + +SIP changes +----------- + * 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, + the username to match in the users list is taken from the Digest header + rather than from the From: field. This feature is considered experimental. + * The "musiconhold" and "musicclass" settings in sip.conf are now removed, + since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4 + * The "localmask" setting was removed in version 1.2 and the reminder about it + being removed is now also removed. + * A new option "busy-level" for setting a level of calls where asterisk reports + a device as busy, to separate it from call-limit + * A new realtime family called "sipregs" is now supported to store SIP registration + data. If this family is defined, "sippeers" will be used for configuration and + "sipregs" for registrations. If it's not defined, "sippeers" will be used for + registration data, as before. + * The SIPPEER function have new options for port address, call and pickup groups + * Added support for T.140 realtime text in SIP/RTP + +IAX2 changes +------------ + * Added the trunkmaxsize configuration option to chan_iax2. + * Added the srvlookup option to iax.conf + * Added support for OSP. The token is set and retrieved through the CHANNEL() + dialplan function. + +DUNDi changes +------------- + * Added the ability to specify arguments to the Dial application when using + the DUNDi switch in the dialplan. + * Added the ability to set weights for responses dynamically. This can be + done using a global variable or a dialplan function. Using the SHELL() + function would allow you to have an external script set the weight for + each response. + +Voicemail Changes +----------------- + * Added the ability to customize which sound files are used for some of the + prompts within the Voicemail application by changing them in voicemail.conf + * Added the ability for the "voicemail show users" CLI command to show users + configured by the dynamic realtime configuration method. + Miscellaneous ------------- @@ -53,11 +128,9 @@ Miscellaneous * 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. - * Added the trunkmaxsize configuration option to chan_iax2. * Added G729 passthrough support to chan_phone for Sigma Designs boards. * Added the parkedcalltransfers option to features.conf * Added 's' option to Page application. - * Added the srvlookup option to iax.conf * Added 'E' and 'V' commands to ExternalIVR. * Added 'DBDel' and 'DBDelTree' manager commands. * Added 'o' and 'X' options to Chanspy. @@ -68,71 +141,3 @@ Miscellaneous * Added a new realtime configuration module, res_config_sqlite * Added a new dialplan application, Bridge, which allows you to bridge the calling channel to any other active channel on the system. - -AMI - The manager (TCP/TLS/HTTP) --------------------------------- - * Added the URI redirect option for the built-in HTTP server - * The output of CallerID in Manager events is now more consistent. - CallerIDNum is used for number and CallerIDName for name. - * enable https support for builtin web server. - See configs/http.conf.sample for details. - * Added a new action, GetConfigJSON, which can return the contents of an - Asterisk configuration file in JSON format. This is intended to help - improve the performance of AJAX applications using the manager interface - over HTTP. - * SIP and IAX manager events now use "ChannelType" in all cases where we - indicate channel driver. Previously, we used a mixture of "Channel" - and "ChannelDriver" headers. - * Added a "Bridge" action which allows you to bridge any two channels that - are currently active on the system. - -Dialplan functions ------------------- - * Added the DEVSTATE() dialplan function which allows retrieving any device - 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 - -CLI Changes ------------ - * New CLI command "core show settings" - * Added 'core show channels count' CLI command. - -SIP changes ------------ - * 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, - the username to match in the users list is taken from the Digest header - rather than from the From: field. This feature is considered experimental. - * The "musiconhold" and "musicclass" settings in sip.conf are now removed, - since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4 - * The "localmask" setting was removed in version 1.2 and the reminder about it - being removed is now also removed. - * A new option "busy-level" for setting a level of calls where asterisk reports - a device as busy, to separate it from call-limit - * A new realtime family called "sipregs" is now supported to store SIP registration - data. If this family is defined, "sippeers" will be used for configuration and - "sipregs" for registrations. If it's not defined, "sippeers" will be used for - registration data, as before. - * The SIPPEER function have new options for port address, call and pickup groups - * Added support for T.140 realtime text in SIP/RTP - -DUNDi changes -------------- - * Added the ability to specify arguments to the Dial application when using - the DUNDi switch in the dialplan. - * Added the ability to set weights for responses dynamically. This can be - done using a global variable or a dialplan function. Using the SHELL() - function would allow you to have an external script set the weight for - each response. - -Voicemail Changes ------------------ - * Added the ability to customize which sound files are used for some of the - prompts within the Voicemail application by changing them in voicemail.conf - * Added the ability for the "voicemail show users" CLI command to show users - configured by the dynamic realtime configuration method. |