The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.5.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.5.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
New Features made in this release:
-----------------------------------
* ASTERISK-29446 - app_confbridge: New ConfKick application
(Reported by N A)
* ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
be suppressed
(Reported by N A)
* ASTERISK-29431 - Minimum and maximum dialplan functions
(Reported by N A)
* ASTERISK-29439 - func_volume: Volume function can't be read
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
up during application execution
(Reported by N A)
* ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
domain name
(Reported by George Joseph)
* ASTERISK-29441 - Core reload making TCP endpoints go offline
(Reported by Luke Escude)
* ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
happens when unsubscribe an application from an event source
(Reported by Lucas Tardioli Silveira)
* ASTERISK-28393 - Multidomain support issue
(Reported by
Andrea Sannucci)
* ASTERISK-29433 - res_rtp_asterisk: Server reflexive
candidates use incorrect raddr for RTCP
(Reported by
Chris)
* ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
UASs
(Reported by George Joseph)
* ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
in PJSIP NOTIFY event: dialog XML body
(Reported by Marco
Paland)
* ASTERISK-29370 - chan_sip does not recognize
application/hook-flash
(Reported by N A)
* ASTERISK-29377 - cpool_release_pool "double free or
corruption (out)"
(Reported by Robert Sutton)
* ASTERISK-29372 - file.c switch does not account for flash
events
(Reported by N A)
* ASTERISK-29358 - chan_pjsip: Trace message for progress is
output even if frame is not queued
(Reported by Michael
Maier)
* ASTERISK-29407 - chan_local: Filtering audio formats should
not occur on removed streams
(Reported by Joshua C. Colp)
* ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
wrong SSRC) gets inserted when switching from progress to
established
(Reported by Matthias Hensler)
Improvements made in this release:
-----------------------------------
* ASTERISK-29450 - Allow setting channel variables using
Originate application
(Reported by N A)
* ASTERISK-29459 - Missing configuration from PJSIP to SIP
conversion script
(Reported by N A)
* ASTERISK-29460 - Recognize application/hook-flash in PJSIP
(Reported by N A)
* ASTERISK-29434 - Asterisk reveals pjproject version in STUN
packets
(Reported by Jeremy Lainé)
* ASTERISK-29349 - Silent voicemail option is not completely
silent
(Reported by N A)
* ASTERISK-29380 - Add Flash AMI event to handle flash events
(Reported by N A)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0-rc1
Thank you for your continued support of Asterisk!
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