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! > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
