[asterisk-dev] Asterisk 18.5.0-rc1 Now Available

[Michael Maier]

Hi,

I faced a problem today regarding a conference. One user wasn't able to
enter the conference - after / while entering the second digit of the
conference PIN, the call always was ended repeatedly by the callers
device (Q.850 cause 16). He tried about 5 times. On Asterisk side, I
never could see any dtmf logged.

The exactly same call from the same device / phone number has been
working fine last Thursday w/ Asterisk 18.4.

Unfortunately, I can't reproduce it with any other device here.

I compared signaling from today with the working one - couldn't see any
difference. The RTP stream is not logged at all because of privacy and
it's encrypted anyway.

I'm not sure if it's really an Asterisk problem - but it's somehow
suspicious. Do you by chance have any idea what could have been happened
here regarding your changes?


Thanks
Michael


On 17.06.21 at 17:15 Asterisk Development Team wrote:
> 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!
> 
>