The release of Asterisk 14.5.0 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:
Bugs fixed in this release:
-----------------------------------
| chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable | (Reported by Stefan Engström) | |
| res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 | (Reported by Javier Riveros ) | |
| Duplicate logging in queue log for EXITEMPTY events | (Reported by Ove Aursand) | |
| res_pjsip_session: INVITE retransmissions could still setup the same call again. | (Reported by Richard Mudgett) | |
| res_rtp_asterisk: One way audio when transcoding | (Reported by Henning Holtschneider) | |
| tcptls: Incorrect OpenSSL function call leads to misleading error report | (Reported by Bob Ham) | |
| Crash in Manager Reload when TLS Config Changes | (Reported by Joshua Elson) | |
| [patch]cel_odbc sometimes inserts CEL with wrong eventtime | (Reported by Etienne Lessard) | |
| func_cdr: CDR function does not permit empty values to be assigned | (Reported by gkloepfer) | |
| [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. | (Reported by Frederic LE FOLL) | |
| Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used | (Reported by Corey Farrell) | |
| Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) | (Reported by Evers Lab) | |
| chan_sip: tcpbind uses wrong source address | (Reported by Ksenia) | |
| res_pjsip: Deadlock in T.38 framehook | (Reported by Richard Mudgett) | |
| res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. | (Reported by Richard Mudgett) | |
| SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. | (Reported by Andreas Krüger) | |
| func_speex: Crash caused by frame with no datalen | (Reported by Richard Kenner) | |
| chan_sip: ACK with SDP does not update a direct media bridge | (Reported by Jean Aunis - Prescom) | |
| pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux | (Reported by abelbeck) | |
| pjsip: Add database tables for RLS | (Reported by Joshua Colp) | |
| Asterisk crash if hep.conf have some missing parameters | (Reported by Joel Vandal) | |
| STUN server with non-default-route transport causes INVITE delay | (Reported by George Joseph) | |
| res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) | (Reported by scgm11) | |
| res_rtp_asterisk: Crash when freeing RTCP address string | (Reported by Niklas Larsson) | |
| res_rtp_asterisk: Crash in pjnath when receiving packet | (Reported by Adagio) | |
| format_wav: wav16 format read file only by 320 - half of frame | (Reported by Vitaly K) | |
| format_ogg_vorbis: Memory leak using OGG in MixMonitor | (Reported by Ivan Myalkin) | |
| STUN never works when asterisk started without internet access | (Reported by Jeremy Kister) | |
| Audible clicks when playing sox encoded au file with STREAM FILE AGI command | (Reported by Roman S.) | |
| [UBSAN] strings.h:signed integer overflow in ast_str_case_hash | (Reported by Badalian Vyacheslav) | |
| res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport | (Reported by Richard Begg) | |
| Listening TCP/TLS sockets stop when temporarily out of open files | (Reported by Walter Doekes) | |
| pjsip: Add database tables for PUBLISH support | (Reported by Joshua Colp) | |
| pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). | (Reported by Alexander Traud) | |
| pjproject_bundled: Merge 3 upstream deadlock patches into bundled | (Reported by Ross Beer) | |
| chan_sip: Security vulnerability with client code header | (Reported by Alex VillacÃs Lasso) | |
| Unused realtime MOH classes not purged on 'moh reload' | (Reported by Sébastien Couture) | |
| res_pjsip: Excessive refcount reached on transport ao2 object | (Reported by Ross Beer) | |
| SIP Failed to parse multiple Supported: headers | (Reported by Olle Johansson) | |
| chan_sip: Session Timers required but refused wrongly. | (Reported by Alexander Traud) | |
| res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code | (Reported by Yaacov Akiba Slama) | |
| Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT | (Reported by twisted) | |
| libasteriskssl.so not found when asterisk is installed for the 1st time | (Reported by George Joseph) | |
| xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client | (Reported by Marcello Ceschia) | |
| [patch]SDP crypto tag is validated incorrectly | (Reported by Joerg Sonnenberger) | |
| res_musiconhold: format option is not documented adequately | (Reported by Jens Bürger) | |
| No core dumps because of res_musiconhold chdir. | (Reported by Walter Doekes) | |
| xmpp: starttls problem causes connection spew | (Reported by Matthias Urlichs) | |
| pjproject_bundled build fails to download pjproject source when using cURL | (Reported by Gergely Dömsödi) | |
| JABBER_STATUS fails with improper code 7 for unavailable clients | (Reported by Anthony Critelli) | |
| Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available | (Reported by Jeremy Kister) | |
| WARNING for "JABBER: socket read error" should be more specific | (Reported by Sean Darcy) | |
| rtp_engine: Allocate RTP payloads on a per-session basis | (Reported by Joshua Colp) | |
| cdr: Problem setting variables in h exten | (Reported by scgm11) | |
| app_mixmonitor: Recording out of sync when 183 but no RTP | (Reported by Aaron An) |
Improvements made in this release:
-----------------------------------
| Investigate heavy memory utilization by res_pjsip_pubsub | (Reported by Richard Mudgett) | |
| res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip | (Reported by Nir Simionovich (GreenfieldTech - Israel)) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0
Thank you for your continued support of Asterisk!
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-announce mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-announce
|