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	Update CHANGES and UPGRADE.txt for 18.12.0
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							| @@ -12,6 +12,70 @@ | ||||
| === | ||||
| ============================================================================== | ||||
|  | ||||
| ------------------------------------------------------------------------------ | ||||
| --- Functionality changes from Asterisk 18.11.3 to Asterisk 18.12.0 ---------- | ||||
| ------------------------------------------------------------------------------ | ||||
|  | ||||
| app_confbridge | ||||
| ------------------ | ||||
|  * Added the hear_own_join_sound option to the confbridge user profile to | ||||
|    control who hears the sound_join audio file. When set to 'yes' the user | ||||
|    entering the conference and the participants already in the conference | ||||
|    will hear the sound_join audio file. When set to 'no' the user entering | ||||
|    the conference will not hear the sound_join audio file, but the | ||||
|    participants already in the conference will hear the sound_join audio file. | ||||
|  | ||||
| app_queue | ||||
| ------------------ | ||||
|  * The m option now allows an override music on hold | ||||
|    class to be specified for the Queue application | ||||
|    within the dialplan. | ||||
|  | ||||
| chan_dahdi | ||||
| ------------------ | ||||
|  * Previously, cadences were appended on dahdi restart, | ||||
|    rather than reloaded. This prevented cadences from | ||||
|    being updated and maxed out the available cadences | ||||
|    if reloaded multiple times. This behavior is fixed | ||||
|    so that reloading cadences is idempotent and cadences | ||||
|    can actually be reloaded. | ||||
|  | ||||
| chan_pjsip | ||||
| ------------------ | ||||
|  * added global config option "allow_sending_180_after_183" | ||||
|  | ||||
|    Allow Asterisk to send 180 Ringing to an endpoint | ||||
|    after 183 Session Progress has been send. | ||||
|    If disabled Asterisk will instead send only a | ||||
|    183 Session Progress to the endpoint. | ||||
|  | ||||
|  * Hook flash events can now be sent on a PJSIP channel | ||||
|    if requested to do so. | ||||
|  | ||||
| chan_sip | ||||
| ------------------ | ||||
|  * Session timers get removed on UPDATE | ||||
|    Fix if Asterisk receives a SIP REFER with Session-Timers UAC | ||||
|    that Asterisk maintains Session-Timers when sending UPDATE request | ||||
|  | ||||
| cli | ||||
| ------------------ | ||||
|  * A new CLI command 'dialplan eval function' has been | ||||
|    added which allows users to test the behavior of | ||||
|    dialplan function calls directly from the CLI. | ||||
|  | ||||
| func_db | ||||
| ------------------ | ||||
|  * The function DB_KEYCOUNT has been added, which | ||||
|    returns the cardinality of the keys at a specified | ||||
|    prefix in AstDB, i.e. the number of keys at a | ||||
|    given prefix. | ||||
|  | ||||
| func_evalexten | ||||
| ------------------ | ||||
|  * This adds the EVAL_EXTEN function which may be | ||||
|    used to evaluate data at dialplan extensions. | ||||
|  | ||||
| ------------------------------------------------------------------------------ | ||||
| --- Functionality changes from Asterisk 18.11.1 to Asterisk 18.11.2 ---------- | ||||
| ------------------------------------------------------------------------------ | ||||
|   | ||||
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