The release of Asterisk 18.5.0 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:
New Features made in this release:
-----------------------------------
| app_confbridge: New ConfKick application | (Reported by N A) | |
| app_confbridge: Allow ConfBridge answer to be suppressed | (Reported by N A) | |
| Minimum and maximum dialplan functions | (Reported by N A) | |
| func_volume: Volume function can't be read | (Reported by N A) |
Bugs fixed in this release:
-----------------------------------
| SayNumber triggers WARNING if caller hangs up during application execution | (Reported by N A) | |
| Consolidate res_pjsip_messaging fixes for domain name | (Reported by George Joseph) | |
| Core reload making TCP endpoints go offline | (Reported by Luke Escude) | |
| "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source | (Reported by Lucas Tardioli Silveira) | |
| Multidomain support issue | (Reported by Andrea Sannucci) | |
| res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP | (Reported by Chris) | |
| pjsip: Asterisk isn't tolerant of RFC8760 UASs | (Reported by George Joseph) | |
| [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body | (Reported by Marco Paland) | |
| chan_sip does not recognize application/hook-flash | (Reported by N A) | |
| cpool_release_pool "double free or corruption (out)" | (Reported by Robert Sutton) | |
| file.c switch does not account for flash events | (Reported by N A) | |
| chan_pjsip: Trace message for progress is output even if frame is not queued | (Reported by Michael Maier) | |
| chan_local: Filtering audio formats should not occur on removed streams | (Reported by Joshua C. Colp) | |
| 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:
-----------------------------------
| Allow setting channel variables using Originate application | (Reported by N A) | |
| Missing configuration from PJSIP to SIP conversion script | (Reported by N A) | |
| Recognize application/hook-flash in PJSIP | (Reported by N A) | |
| Asterisk reveals pjproject version in STUN packets | (Reported by Jeremy Lainé) | |
| Silent voicemail option is not completely silent | (Reported by N A) | |
| Add Flash AMI event to handle flash events | (Reported by N A) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
Thank you for your continued support of Asterisk!
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