The release of Certified Asterisk 13.18-cert1 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:
Improvements made in this release:
-----------------------------------
| Upgrade bundled PJPROJECT to 2.7 | (Reported by Richard Mudgett) | |
| [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE | (Reported by David J. Pryke) | |
| alembic: Add support for Microsoft SQL server | (Reported by Florian Floimair) | |
| [patch] libsrtp-2.1.x support | (Reported by Alexander Traud) | |
| Enable CHANNEL function to get from and to tag from SIP Headers | (Reported by Andre Nazario) | |
| Google OAuth 2.0 support for XMPP / Motif | (Reported by Andrey) | |
| Support for GMIME 3.0 | (Reported by Tzafrir Cohen) | |
| [patch] app_queue: Add Priority to AMI QueueStatus | (Reported by Niklas Larsson) | |
| [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip | (Reported by Torrey Searle) | |
| res_pjsip: Add DTMF INFO Failback mode | (Reported by Torrey Searle) | |
| [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup | (Reported by Alexei Gradinari) | |
| app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect | (Reported by Alexei Gradinari) | |
| Core/BuildSystem: Add defines to fix build with LibreSSL | (Reported by Guido Falsi) | |
| Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file | (Reported by Guido Falsi) | |
| audiohooks: Remove redundant codec translations when using audiohooks | (Reported by Michael Walton) | |
| libsrtp-2.x.x support | (Reported by Alex) | |
| res_agi: Set audio format for EAGI audio stream | (Reported by John Fawcett) | |
| 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)) | |
| res_pjsip_session: Add support for overlap dialling | (Reported by Richard Begg) | |
| chan_sip: Add rtcp-mux support | (Reported by Sean Bright) | |
| pjsip - Need a command to list active SIP subscriptions | (Reported by Rusty Newton) | |
| Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec | (Reported by Badalian Vyacheslav) | |
| res_calendar_caldav: Add support for gmail | (Reported by Eduardo Scudeller Libardi) | |
| app_controlplayback: Transmit Silence on ControlPlayback pause | (Reported by Mikheili Dautashvili) |
Bugs fixed in this release:
-----------------------------------
| README refers to security documents that do not exist. | (Reported by Corey Farrell) | |
| crash after an invalid rtcp packet from GT48 FXS gateway | (Reported by Tzafrir Cohen) | |
| res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should | (Reported by Vitezslav Novy) | |
| pjsip_options: qualify_frequency sometimes not applied on reload | (Reported by John Bigelow) | |
| CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... | (Reported by Richard Mudgett) | |
| RTP source learning not working with devices that have some clock issues | (Reported by nappsoft) | |
| RTP: Blind transfer direct media scenario results in one way audio. | (Reported by Richard Mudgett) | |
| Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests | (Reported by George Joseph) | |
| res_http_post: Don't require GMIME_MAJOR_VERSION | (Reported by Joshua Colp) | |
| pjsip: TCP connections may not be destroyed | (Reported by Joshua Colp) | |
| res_pjsip_session: RTP instances leak on 488 responses. | (Reported by Corey Farrell) | |
| chan_sip: Security vulnerability with client code header (revisited) | (Reported by Richard Mudgett) | |
| (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines | (Reported by Kim youngsung) | |
| Regression: Deadlock between AOR named lock and pjproject grp lock | (Reported by shaurya jain) | |
| Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more | (Reported by Michael Maier) | |
| res_xmpp: Crash if OAuth 2.0 is used before curl is loaded | (Reported by Ronald Raikes) | |
| ARI: Node ARI client broken in latest versions of 13 and 14 | (Reported by Benjamin Keith Ford) | |
| res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. | (Reported by dtryba) | |
| res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs | (Reported by dtryba) | |
| cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured | (Reported by Tzafrir Cohen) | |
| [patch] app_queue: Music On Hold for real-time queues is not reset to default | (Reported by Nathan Bruning) | |
| Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate | (Reported by Allen Ford) | |
| res_pjsip: Loss of SIP registrations causing unavailable endpoints | (Reported by Richard Mudgett) | |
| res_ari: Memory leaks in ARI when using Content-Type: application/json | (Reported by David Hajek) | |
| chan_sip: tcpbind uses wrong source address | (Reported by Ksenia) | |
| [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS | (Reported by Alexander Traud) | |
| vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. | (Reported by Corey Farrell) | |
| res_pjsip_mwi: uninitialized value from ast_strings_match | (Reported by Corey Farrell) | |
| [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved | (Reported by Benoît Dereck-Tricot) | |
| Status of RFC 3323 and PJSIP | (Reported by dtryba) | |
| app_queue: does its check-makeannouncement-logic twice each head-caller-loop | (Reported by Stefan Engström) | |
| Contact is improperly translated after d178f497 | (Reported by Sean Bright) | |
| Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) | (Reported by Ross Beer) | |
| A codeblock that maintains a bug,but maybe the codeblock will never run | (Reported by Huangyx) | |
| Realtime config fail with PostgreSQL version before 9.1 | (Reported by Rodrigo Ramirez Norambuena) | |
| bridge_native_rtp: half-way direct media when using early bridging | (Reported by Jean Aunis - Prescom) | |
| Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability | (Reported by Ross Beer) | |
| tcptls: Incorrect OpenSSL function call leads to misleading error report | (Reported by Bob Ham) | |
| SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 | (Reported by Marcello Ceschia) | |
| RTCP needs better packet validation to resist port scans. | (Reported by Richard Mudgett) | |
| RTP: One way audio with direct media and strictrtp=yes. | (Reported by Richard Mudgett) | |
| module reload res_calendar.so does not reload everything in calendar.conf | (Reported by Jesper) | |
| res_calendar does not process CalDAV from Owncloud [fix included] | (Reported by Stefan Gofferje) | |
| res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. | (Reported by Jesper) | |
| RTP Multicast of L16 (type 10): Asterisk and wireshark disagree | (Reported by Tzafrir Cohen) | |
| [patch]external_media_address and external_signaling_address don't always honor localnet | (Reported by Walter Doekes) | |
| CDR: CDR(start,u) function won't work in cdr_custom config | (Reported by Jacek Konieczny) | |
| chan_sip: Asterisk crashing when subscription doesn't get set | (Reported by Bryan Walters) | |
| res_smdi: convert to astobj2 | (Reported by Corey Farrell) | |
| SDP origin attribute modified when issuing re-INVITE because of directmedia=yes | (Reported by saghul) | |
| alembic: prune_on_boot fix erroneous | (Reported by Florian Floimair) | |
| When in queue on g722 with interruptions, music on hold can get stuck and no longer play | (Reported by Jens T.) | |
| nat/external_media settings ignored in 14.4.1 | (Reported by Christopher van de Sande) | |
| PJSIP external_media_address ignored if no local_net options are provided | (Reported by Matt Jordan) | |
| Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive | (Reported by Ross Beer) | |
| Crash when freeing dtls_cfg->cafile | (Reported by Richard Kenner) | |
| ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c | (Reported by Tzafrir Cohen) | |
| libc segfault upon entry into app_directory | (Reported by David Moore) | |
| Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash | (Reported by Ross Beer) | |
| core: ast_safe_system command injection possible. | (Reported by Corey Farrell) | |
| res_rtp_asterisk: Media can be hijacked even with strict RTP enabled | (Reported by Joshua Colp) | |
| Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name | (Reported by James Terhune) | |
| app_minivm fails to clean up mkstemp files | (Reported by Walter Doekes) | |
| several filename bugs in Record() application | (Reported by klaus3000) | |
| alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table | (Reported by Florian Floimair) | |
| ControlPlayback fails to play files with names containing certain non-alpha characters | (Reported by Jonathan White) | |
| When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. | (Reported by Jim Van Meggelen) | |
| When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored | (Reported by Eelco Brolman) | |
| [patch] app_queue: Wrong queue stat calculation | (Reported by sungtae kim) | |
| Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used | (Reported by Torrey Searle) | |
| XMPP OAuth not working due to inverted logic | (Reported by Michael Kuron) | |
| res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical | (Reported by Mark Thompson) | |
| If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed | (Reported by Seán C. McCord) | |
| Either asterisk or pjproject isn't re-using tcp connections (again) | (Reported by George Joseph) | |
| IPv6 receive address in message doesn't include brackets | (Reported by Scott Griepentrog) | |
| [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used | (Reported by Torrey Searle) | |
| RTP session is not fully destroyed on channel hangup | (Reported by Matt Jordan) | |
| Asymmetric codecs when asymmetric_rtp_codec=no | (Reported by Jesse Ross) | |
| Asterisk 15.0.0-Beta1 does not compile | (Reported by Ira Emus) | |
| res_pjsip: PJSIP presence - missing braces around the status element in XML | (Reported by Abraham Liebsch) | |
| Asterisk won't compile on Fedora 26 with devmode enabled. | (Reported by Corey Farrell) | |
| Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly | (Reported by Sergej Kasumovic) | |
| say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. | (Reported by Nicolas Riendeau) | |
| sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 | (Reported by Corey Farrell) | |
| app_playback.c:say_date_generic use timezonename parameter | (Reported by Holger Hans Peter Freyther) | |
| res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use | (Reported by Joshua Colp) | |
| [patch]res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media | (Reported by Dan Jenkins) | |
| confbridge: Name recordings are left on filesystem | (Reported by Sergej Kasumovic) | |
| chan_iax2: On reload MWI taskprocessors keep adding up | (Reported by Sergej Kasumovic) | |
| sounds: New 3-D Binaural audio features require new sound prompts | (Reported by Rusty Newton) | |
| French conf-adminmenu, conf-usermenu prompts differ in content from the English files | (Reported by Benoit Duverger) | |
| Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) | (Reported by Rusty Newton) | |
| configs: Erroneous load directive in sample configuration results in "Error loading module 'res_pjsip_multihomed.so'" | (Reported by HZMI8gkCvPpom0tM) | |
| [patch]core: when setting 'maxfiles' in asterisk.conf, a message is printed, even in rasterisk -x | (Reported by Tzafrir Cohen) | |
| res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@' | (Reported by Maxim Vasilev) | |
| res_rtp_asterisk: Deadlock when TURN session in use | (Reported by Jatin Jain) | |
| Crash using 'data get' CLI command | (Reported by Sean Bright) | |
| [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. | (Reported by Alexander Traud) | |
| ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting | (Reported by James Terhune) | |
| channel: ast_waitfordigit_full fails to clear flag in an error branch. | (Reported by Corey Farrell) | |
| PJSIP: Deadlock using TCP transport | (Reported by Richard Mudgett) | |
| pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure | (Reported by George Joseph) | |
| chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE | (Reported by George Joseph) | |
| Duplicate logging in queue log for EXITEMPTY events | (Reported by Ove Aursand) | |
| call hangup after leaving app_queue | (Reported by Marek Cervenka) | |
| res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation | (Reported by Joshua Colp) | |
| rtp: Crash in ast_rtp_codecs_payload_code() | (Reported by Ross Beer) | |
| core_local: local channel data not being properly unref'ed and unlocked | (Reported by Kevin Harwell) | |
| bridge: stuck channel(s) after failed attended transfer | (Reported by Kevin Harwell) | |
| app_voicemail reloads result in leaked IMAP sockets. | (Reported by Louis Jocelyn Paquet) | |
| res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact | (Reported by Alexei Gradinari) | |
| res_stasis: Stolen channel references are leaking | (Reported by George Joseph) | |
| Comment typo format_g729.c | (Reported by Matthew Fredrickson) | |
| res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip | (Reported by Zach R) | |
| res_corosync segfaults at startup with corosync version > 2.x | (Reported by mdu113) | |
| res_ari: Crash when no ari.conf configuration file exists | (Reported by Ronald Raikes) | |
| Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times. | (Reported by Chris Howard) | |
| Core/PBX: [patch] Deadlock between dialplan execution and application unregistration | (Reported by Frederic LE FOLL) | |
| Seg Fault in ast_sorcery_object_get_id at sorcery.c | (Reported by Ryan Smith) | |
| res_pjsip_transport_websocket: segfault in get_write_timeout | (Reported by Jørgen H) | |
| res_rtp_asterisk: Incorrect SSRC change for RTCP component | (Reported by Michael Walton) | |
| bridging: T.38 request is lost when channels are added to bridge | (Reported by Torrey Searle) | |
| res_pjsip_refer/session: Calls dropped during transfer | (Reported by Kevin Harwell) | |
| Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network | (Reported by alex) | |
| chan_pjsip: Device state is idle when channel from endpoint is in early media | (Reported by Joshua Colp) | |
| chan_pjsip: Flipping between codecs | (Reported by Michael Maier) | |
| chan_pjsip would send INVITE to 'Unreachable' endpoints | (Reported by Jacek Konieczny) | |
| bridge: Crash when freeing frame and snooping | (Reported by Michel R. Vaillancourt) | |
| Background in realtime | (Reported by Andrew Nowrot) | |
| channel / meetme: Fix missing parentheses | (Reported by Joshua Colp) | |
| GET /recordings/stored returns 500 Internal Server Error | (Reported by Tim Morgan) | |
| [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec | (Reported by Frankie Chin) | |
| Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded | (Reported by Tzafrir Cohen) | |
| srtp's crypto_get_random deprecated | (Reported by Tzafrir Cohen) | |
| AGI - RECORD FILE - documentation doesn't describe BEEP argument | (Reported by Rusty Newton) | |
| Async AGI crashes Asterisk when issuing "set variable" command without args | (Reported by Antoine Pitrou) | |
| Malformed AGI 520 Usage response | (Reported by Tony Mountifield) | |
| DTLS configuration can not be specified in the general section - documentation | (Reported by Ben Langfeld) | |
| res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space | (Reported by John Harris) | |
| app_queue: Agent not called when caller is parked | (Reported by wushumasters) | |
| app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime | (Reported by Etienne Lessard) | |
| app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel | (Reported by David Brillert) | |
| app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call | (Reported by Lorne Gaetz) | |
| app_confbridge: ConfBridge sometimes does not play user name recording while leaving | (Reported by Robert Mordec) | |
| res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 | (Reported by Javier Riveros ) | |
| chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable | (Reported by Stefan Engström) | |
| res_pjsip_session: Wrong From on reinvite when request and To URI differ | (Reported by Yasin CANER) | |
| Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP | (Reported by Sandro Gauci) | |
| Out of bound memory access in PJSIP multipart parser crashes Asterisk | (Reported by Sandro Gauci) | |
| Asterisk Skinny memory exhaustion vulnerability leads to DoS | (Reported by Sandro Gauci) | |
| Audit manipulation of channel flags without locks | (Reported by Joshua Colp) | |
| res_pjsip_session: INVITE retransmissions could still setup the same call again. | (Reported by Richard Mudgett) | |
| Problems with Blind Transfer, PJSIP (Aastra 6869i) | (Reported by Matthias Binder) | |
| 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) | |
| res_rtp_asterisk: One way audio when transcoding | (Reported by Henning Holtschneider) | |
| 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) | |
| 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) | |
| 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) | |
| func_speex: Crash caused by frame with no datalen | (Reported by Richard Kenner) | |
| 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 Sebastian Gutierrez) | |
| 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.) | |
| 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) | |
| [UBSAN] strings.h:signed integer overflow in ast_str_case_hash | (Reported by Badalian Vyacheslav) | |
| 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) | |
| xmpp: starttls problem causes connection spew | (Reported by Matthias Urlichs) | |
| res_musiconhold: format option is not documented adequately | (Reported by Jens Bürger) | |
| No core dumps because of res_musiconhold chdir. | (Reported by Walter Doekes) | |
| 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) | |
| cdr: Problem setting variables in h exten | (Reported by Sebastian Gutierrez) | |
| res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets | (Reported by Max Norba) | |
| res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. | (Reported by Vinod Dharashive) | |
| res_pjsip_pubsub: Crash when generating xpidf content | (Reported by Andrew Green) | |
| Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled | (Reported by Kirsty Tyerman) | |
| app_mixmonitor: Recording out of sync when 183 but no RTP | (Reported by Aaron An) | |
| app_queue: Queue stops calling members with local interface after forwarding in previous call | (Reported by Robert Mordec) | |
| res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome | (Reported by Dan Jenkins) | |
| autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). | (Reported by Krzysztof Trempala) | |
| res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension | (Reported by Torrey Searle) | |
| core: Malformed pattern matching extension (various factors) results in crash | (Reported by xrobau) | |
| chan_iax2: Reload of iax peer results in loss of host address/port | (Reported by Richard Begg) | |
| Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) | (Reported by Matt Jordan) | |
| Document the fact that Asterisk HEP support only works with the PJSIP channel driver | (Reported by Olivier Krief) | |
| Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk | (Reported by Roman Bedros) | |
| stasis_cache.c:845 caching_topic_exec: - misleading ERROR message | (Reported by Smirnov Aleksey) | |
| chan_pjsip: Dialplan function race condition | (Reported by Joshua Colp) | |
| chan_sip: Call not cancelled after receiving a 422 response | (Reported by Jean Aunis - Prescom) | |
| pjsip/cli_commands: pjsip show channelstats shows wrong codec | (Reported by Kevin Harwell) | |
| res_pjsip: Crash when using IPv6 and Transport ws,wss | (Reported by Michael Balen) | |
| app_voicemail: Cannot set fromstring on a per-mailbox basis | (Reported by Mark Scholten) | |
| Saynumber is trying to get "and" from "digits/" subfolder | (Reported by Jonathan Harris) | |
| Long lines in call files cause spurious syntax error | (Reported by Dave Olszewski) | |
| res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' | (Reported by Jørgen H) | |
| res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging | (Reported by Dmitry Wagin) | |
| pjsip.conf.sample: user_agent: still refers to branch 12 | (Reported by Tzafrir Cohen) | |
| PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist | (Reported by Mark Michelson) | |
| res_pjsip: Crash when calling PJSIPShowEndpoint | (Reported by Jørgen H) | |
| res_pjsip_outbound_registration doesn't know about network change events | (Reported by George Joseph) | |
| chan_sip : Asterisk restart seems to be required for changing encryption option | (Reported by benasse) | |
| bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio | (Reported by Sean Bright) | |
| res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication | (Reported by Peter Sokolov) | |
| [patch] Fix download_externals To Allow The Use Of curl Or wget | (Reported by Michael L. Young) | |
| Pattern matching with res_config_mysql extensions does not behave as expected | (Reported by Charlie Smurthwaite) | |
| PJSIP Segfault 13.13.1 (Bundled PJSIP) | (Reported by Nic Colledge) | |
| [patch] DUNDi weight parameter not processed correctly | (Reported by Peter Racz) | |
| [patch] Error during LDAP modify action when user unregisters | (Reported by Nicholas John Koch) | |
| res_pjsip: Using an auth object for inbound and outbound authentication fails. | (Reported by Richard Mudgett) | |
| Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c | (Reported by Michael Maier) | |
| Function vmauthenticate accesses uninitialized memory | (Reported by Filip Jenicek) | |
| [patch] Integrity Check Of PJSIP Download Fails | (Reported by Michael L. Young) | |
| [patch] Fix query with double backslash in string literals and stop log warnings | (Reported by Humberto Figuera) | |
| res_config_sqlite3 uses incorrect query - unnecessary escape | (Reported by Stepan) | |
| SQlite3: Realtime queue loading fails after PRAGMA query result | (Reported by Scott Griepentrog) | |
| http: Crash on Reload Only in ast_tcptls_server_start | (Reported by Joshua Elson) | |
| Phone default have not ringing on ARM | (Reported by Igor Goncharovsky) | |
| pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh | (Reported by Zach R) | |
| res_pjsip_mwi: Asterisk does not terminate MWI subscription | (Reported by Carl Fortin) | |
| Asterisk fails building with OpenSSL 1.1.0 | (Reported by Tzafrir Cohen) | |
| VoiceMailPlayMsg not playing messages via realtime | (Reported by Ryan Rittgarn) | |
| [patch] 'Silence' is truncated in Record() | (Reported by var) | |
| chan_pjsip: Error when calling PJSIP client with domain specified | (Reported by Norbert Varga) | |
| core: Protect flags during ast_waitfor | (Reported by Joshua Colp) | |
| pbx: AMI Originate ignore "failed" extension on call failure | (Reported by Nasir Iqbal) | |
| configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample | (Reported by Torrey Searle) | |
| Crash in srv.c on startup with pjsip | (Reported by nappsoft) | |
| res_stasis_device_state: Duplicate subscriptions when multiple received at same time | (Reported by Joshua Colp) | |
| res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. | (Reported by Anthony Messina) | |
| MixMonitorMute mutes through stream if already slinear (e.g. Originate) | (Reported by David Woolley) | |
| ari: Channels with pre-dial handlers cannot be hung up via ARI | (Reported by Tom Pawelek) | |
| core: Possibility of a frame "imbalance" leading to stuck channels. | (Reported by Mark Michelson) | |
| res_agi: run_agi eats frames it shouldn't | (Reported by George Joseph) | |
| ASTERISK-25951 causes issues for callerid manipulation through agi | (Reported by Morten Tryfoss) | |
| Crash on invalid contact domain (pjsip aor) | (Reported by Dmitriy) | |
| res_pjsip: Assertion when sending OPTIONS request to endpoint | (Reported by Ross Beer) | |
| build_tools: make_build_h does not handle \ in user name | (Reported by Kirill Katsnelson) | |
| AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" | (Reported by Kirill Katsnelson) | |
| app_queue: Random queues disappear on "core reload queue all" | (Reported by Kirill Katsnelson) | |
| res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect | (Reported by Michael Maier) | |
| res_pjsip_endpoint_identifier_ip: Add support for SRV | (Reported by Joshua Colp) | |
| PJPROJECT: Detecting compiled max log level does not work. | (Reported by Richard Mudgett) | |
| voicemail API test: uses varlibdir instead of datadir for a sound file | (Reported by Tzafrir Cohen) | |
| voicemail API test: confuses expected and actual values | (Reported by Tzafrir Cohen) | |
| res_sorcery_memory_cache: memory leak on every sorcery memory cache populate | (Reported by Ustinov Artem) | |
| [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 | (Reported by Aaron An) | |
| Crash when setting remote address on RTP instance | (Reported by Richard Mudgett) | |
| [patch] Outgoing SIP-URI Dialing via PJSIP | (Reported by Alexander Traud) | |
| Remember SDP negotiation on SIP_CODEC_INBOUND. | (Reported by Alexander Traud) | |
| chan_pjsip: Crash when using CHANNEL dialplan function around masquerade | (Reported by Joshua Colp) | |
| res_pjsip: Various issues with compact SIP headers | (Reported by Joshua Elson) | |
| [patch]pjsip: Transfers Broken with Compact Headers Enabled | (Reported by JoshE) | |
| app_queue: Queue application does not ring members with Local interface | (Reported by Jonas Kellens) | |
| chan_sip: Segfaults upon reload if client with MWI wasn't registered | (Reported by Michael Kuron) | |
| build: GCC 5.1.x catches some new const, array bounds and missing paren issues | (Reported by George Joseph) | |
| Need more explicit debug when PJSIP dialstring is invalid | (Reported by Rusty Newton) | |
| Message.c: Message channel becomes saturated with frames leading to spammy log messages | (Reported by Jonathan Rose) | |
| pjproject_bundled doesn't verify already downloaded tarballs | (Reported by George Joseph) | |
| chan_sip: Allows To-tag checks to be bypassed, setting up new calls | (Reported by Walter Doekes) | |
| codec_opus: Recursiveness when parsing fmtp line | (Reported by Jørgen H) | |
| PJSIPShowRegistrationsInbound just dumps all aors | (Reported by George Joseph) | |
| res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" | (Reported by Juris Breicis) | |
| res_rtp_asterisk: Can't bind on systems without IPv6 | (Reported by Guido Falsi) | |
| Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 | (Reported by Marek Cervenka) | |
| mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' | (Reported by Tzafrir Cohen) | |
| res_rtp_asterisk: RTT miscalculation in RTCP | (Reported by Hector Royo Concepcion) | |
| chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. | (Reported by Michael Kuron) | |
| [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no | (Reported by Alexei Gradinari) | |
| chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression | (Reported by Michael Keuter) | |
| app_voicemail: Asterisk crashes when MailboxExists is used | (Reported by Doug Lytle) |
New Features made in this release:
-----------------------------------
| PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. | (Reported by Richard Mudgett) | |
| [patch]AMI : Add CancelAtxfer Action | (Reported by Thomas Sevestre) | |
| core: Add support for timelen parsing to ast_parse_arg and ACO. | (Reported by Corey Farrell) | |
| func_channel: Add ability to get the callid so dialplan has access to it. | (Reported by Richard Mudgett) | |
| res_pjsip: Add endpoint identification scheme based on a configured SIP header/value | (Reported by Matt Jordan) | |
| [patch] Allow "Comedian Mail" branding to be removed | (Reported by John Covert) | |
| Make logging PJPROJECT messages a bit easier | (Reported by Richard Mudgett) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.18-cert1
Thank you for your continued support of Asterisk!
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