The release of Certified Asterisk 18.9-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:
Deprecations made in this release:
-----------------------------------
| app_meetme: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| app_osploop: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_alsa: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_mgcp: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_skinny: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| res_pktccops: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| cdr_mysql: Deprecated in 1.8, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_mysql: Deprecated in 1.8, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_ices: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_macro: Deprecated in 16, to be removed in 21 | (Reported by Joshua C. Colp) | |
| app_fax: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_url: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_image: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_nbscat: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| app_dahdiras: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| cdr_syslog: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| chan_oss: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| chan_phone: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| chan_sip: Deprecated in 17, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_nbs: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| chan_misdn: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| chan_vpb: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| res_config_sqlite: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| res_monitor: Deprecated in 16, to be removed in 21 | (Reported by Joshua C. Colp) | |
| conf2ael: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) | |
| muted: Deprecated in 16, to be removed in 19 | (Reported by Joshua C. Colp) |
Security bugs fixed in this release:
-----------------------------------
| res_stir_shaken: Blind SSRF vulnerabilities | (Reported by Clint Ruoho) | |
| res_stir_shaken: Resource exhaustion with large files | (Reported by Benjamin Keith Ford) | |
| ${SQL_ESC()} not correctly escaping a terminating \ | (Reported by Leandro Dardini) | |
| pjproject: Security fixes for things | (Reported by Kevin Harwell) | |
| Crash in PJSIP TLS transport | (Reported by Andrew Yager) | |
| chan_pjsip: Remote denial of service by an authenticated user | (Reported by Ivan Poddubny) | |
| ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash | (Reported by Gregory Massel) | |
| sRTP Replay Protection ignored; even tears down long calls | (Reported by Alexander Traud) | |
| res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash | (Reported by Ivan Poddubny) | |
| res_pjsip_diversion: Crash if Tel URI contains History-Info | (Reported by Torrey Searle) | |
| pjsip: Crash on call rejection during high load | (Reported by Sandro Gauci) | |
| chan_sip: Depending on configuration an INVITE can alter Addr of a peer | (Reported by Andrey V. T.) | |
| Bypass SYSTEM write permission in manager action allows system commands execution | (Reported by Eliel Sardañons) | |
| res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash | (Reported by Alexei Gradinari) | |
| res_pjsip_messaging: In-dialog MESSAGE with no body causes crash | (Reported by Gil Richard) | |
| Broken SDP can cause a segfault in a T.38 reINVITE | (Reported by Francesco Castellano) | |
| Asterisk segfault when rtp negotiation is wrong or fails | (Reported by Sotiris Ganouris) | |
| Buffer overflow for DNS SRV/NAPTR records | (Reported by Jan Hoffmann) | |
| res_http_websocket: Crash when reading HTTP Upgrade requests | (Reported by Sean Bright) |
New Features made in this release:
-----------------------------------
| res_tonedetect: Add call progress tone detection | (Reported by N A) | |
| [patch] app_queue Add Login Time and Last Paused Times to Queue Members | (Reported by Jamuel Starkey) | |
| Add CHANNEL_EXISTS function | (Reported by N A) | |
| Add SendMF application | (Reported by N A) | |
| Add STRBETWEEN function | (Reported by N A) | |
| Add file and directory functions | (Reported by N A) | |
| Add SAYFILES function | (Reported by N A) | |
| Add tone detection module | (Reported by N A) | |
| Option for Read to be able to accept # | (Reported by Sta Retji) | |
| Add audio scrambler | (Reported by N A) | |
| Function to drop frames in the TX or RX directions | (Reported by N A) | |
| Add PJSIP_HEADERS() and ability to read header by pattern | (Reported by Igor Goncharovsky) | |
| AGI channel_status failure | (Reported by bbawkon) | |
| Function to asynchronously store digits dialed | (Reported by N A) | |
| New application to reload modules | (Reported by N A) | |
| Add application to wait for condition | (Reported by N A) | |
| app_dial: Expand A option to allow announcement playback to caller | (Reported by N A) | |
| 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) | |
| Chan_pjsip does not support unauthenticated OPTIONS ping | (Reported by Ross Beer) | |
| Implement support for History-Info | (Reported by Torrey Searle) | |
| [patch] allow Asterisk to set high ToS bits as non-root on Linux | (Reported by Matt Addison) | |
| CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything | (Reported by candrews) | |
| res_pjsip_endpoint_identifier_ip: Add ability to match on source port | (Reported by Sean Bright) | |
| app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" | (Reported by laszlovl) | |
| func_curl: CURLOPT cannot set Content-Type header | (Reported by Martin Tomec) | |
| func_jitterbuffer: Add support for video synchronization | (Reported by Joshua C. Colp) | |
| [patch] Unregister a realtime moh class | (Reported by Byron Clark) | |
| Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain | (Reported by Stas Kobzar) | |
| Add native Prometheus support to Asterisk | (Reported by Matt Jordan) | |
| res_pjsip: New configuration setting to allow disabling norefersub | (Reported by Dan Cropp) | |
| Added ARI resource /ari/channels/{channelid}/rtp_statistics | (Reported by sungtae kim) | |
| res_stasis: Add ability to switch applications | (Reported by Benjamin Keith Ford) | |
| add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip | (Reported by Torrey Searle) | |
| res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability | (Reported by Nick French) |
Bugs fixed in this release:
-----------------------------------
| Failed to sign STIR/SHAKEN payload with functionality not enabled | (Reported by Claude Diderich) | |
| VoiceMailMain() fails when encountering non-numeric CALLERID(num) | (Reported by Mark Murawski) | |
| SAY_DTMF_INTERRUPT channel variable is not honored | (Reported by Sean Bright) | |
| Deadlock in bridge_channel_internal_join() on local channels. | (Reported by Krzysztof Trempala) | |
| progdocs: Hidden code sections with syntax errors. | (Reported by Alexander Traud) | |
| progdocs: Fix grouping for latest Doxygen | (Reported by Alexander Traud) | |
| Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning | (Reported by Mario Ban) | |
| stir/shaken: Requires GNU designator | (Reported by Alexander Traud) | |
| chan_misdn: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: doxyref.h outdated | (Reported by Alexander Traud) | |
| xmldoc: Fix for Doxygen | (Reported by Alexander Traud) | |
| Segfault in __ao2_ref if refdebug = yes | (Reported by Alexei Gradinari) | |
| channels: Fix for Doxygen | (Reported by Alexander Traud) | |
| bridging: Infinite loop when both Local channel halves in same bridge | (Reported by Joshua C. Colp) | |
| odbc: Fix for Doxygen | (Reported by Alexander Traud) | |
| parking: Fix for Doxygen | (Reported by Alexander Traud) | |
| frame: Fix for Doxygen | (Reported by Alexander Traud) | |
| res_ari: Fix for Doxygen | (Reported by Alexander Traud) | |
| channel: Fix for Doxygen | (Reported by Alexander Traud) | |
| stasis: Fix for Doxygen | (Reported by Alexander Traud) | |
| app: Fix for Doxygen | (Reported by Alexander Traud) | |
| res_xmpp: Fix for Doxygen | (Reported by Alexander Traud) | |
| addons: Fix for Doxygen. | (Reported by Alexander Traud) | |
| res_pjsip: Fix for Doxygen | (Reported by Alexander Traud) | |
| chan_iax2: Fix for Doxygen | (Reported by Alexander Traud) | |
| bridges: Fix for Doxygen | (Reported by Alexander Traud) | |
| tests: Fix for Doxygen | (Reported by Alexander Traud) | |
| apps: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: Avoid name with Doxygen \file | (Reported by Alexander Traud) | |
| bridge_channel: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: Avoid multiple use of section labels | (Reported by Alexander Traud) | |
| progdocs: Use Doxygen \example correctly | (Reported by Alexander Traud) | |
| app_morsecode: Fix deadlock | (Reported by N A) | |
| res_pjsip_callerid: Fix OLI parsing | (Reported by N A) | |
| app_read: Fix custom terminator functionality regression | (Reported by N A) | |
| BuildSystem: In POSIX sh, == in place of = is undefined. | (Reported by Alexander Traud) | |
| sig_analog: Fix truncated buffer copy | (Reported by N A) | |
| pbx: "dialplan reload" is removing minus symbol from dynamic hints | (Reported by Daniel Zanutti) | |
| VoiceMail does not cancel recording on rerecord hangup | (Reported by N A) | |
| res_snmp: Not build on recent Debian distributions. | (Reported by Alexander Traud) | |
| stasis: Clang 13 warns about the unused but set variable dispatched. | (Reported by Alexander Traud) | |
| aelparse: GCC 11.2 found two maybe uninitialized | (Reported by Alexander Traud) | |
| GCC 11.2: two stringop-overread | (Reported by Alexander Traud) | |
| Squash compiler issues generated by gcc 11 | (Reported by George Joseph) | |
| Using --with-crypto and --with-ssl fails on a recompile | (Reported by George Joseph) | |
| func_talkdetect's logic is completely broken | (Reported by Moritz Fain) | |
| stun: Not all users provide a dst to ast_stun_request | (Reported by Dennis Haney) | |
| make install downloads x86_32 variants of external modules on non Intel architectures | (Reported by Corey Farrell) | |
| [patch] - IAX2 Call Encryption Fails with RSA authentication | (Reported by Michael Munger) | |
| res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it | (Reported by Matthew Kern) | |
| app_read: Fix null pointer crash regression | (Reported by N A) | |
| res_rtp_asterisk: memory leak | (Reported by Jean Aunis - Prescom) | |
| ari: Listing bridges fails when dialing bridge exists | (Reported by Joshua C. Colp) | |
| messaging: AMI MessageSend does not support same parameters as dialplan application | (Reported by Brian J. Murrell) | |
| app_queue: Custom device state using included hints do not update | (Reported by N A) | |
| Build failure when disabling PJSIP support | (Reported by Guido Falsi) | |
| MP3Player don' t work with actual mpg123 versions | (Reported by Carlos Oliva) | |
| pjproject includes trailing whitespace in sdp format attributes | (Reported by George Joseph) | |
| ARI external media channel creation doesn't set option data | (Reported by sungtae kim) | |
| test_abstract_jb: frames leak | (Reported by Corey Farrell) | |
| res_snmp: gcc 11 needs -fPIC to compile correctly | (Reported by George Joseph) | |
| Asterisk is unable to read extended number format terminfo files | (Reported by Sean Bright) | |
| dns: Core ast_dns_get_nameservers does not support configured IPv6 servers | (Reported by Isaac McDonald) | |
| ConfBridge errors on creation conference room | (Reported by Alexander Zharov) | |
| ARI: external media create doesn't use body parameter | (Reported by sungtae kim) | |
| app_agent_pool: XML Doc: unterminated entity reference | (Reported by Alexander Traud) | |
| Subsequent 'ael reload' will cause a lock up | (Reported by Mark Murawski) | |
| app_queue: Core reload resets queue stats, even when keepstats=yes | (Reported by Luke Escude) | |
| res_rtp_asterisk: sqrt(.) requires the header math.h. | (Reported by Alexander Traud) | |
| sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling | (Reported by Sarah Autumn) | |
| res_pjproject: Can't map pjproject log messages to Asterisk TRACE | (Reported by George Joseph) | |
| app_milliwatt: Milliwatt application doesn't use the proper timings | (Reported by N A) | |
| chan_mgcp, resp_pktccops ast_debug support | (Reported by Tomas Maldonado) | |
| aelparse: include of context with timings fails | (Reported by Alexander Traud) | |
| Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) | (Reported by Ernani José Camargo Azevedo) | |
| cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used | (Reported by N A) | |
| statsd: Remove non-standard metric type Meter | (Reported by Rijnhard Hessel) | |
| app_voicemail2 became a bit silent, lately | (Reported by siggi) | |
| G729 audio gets corrupted by Asterisk due to smoother | (Reported by under) | |
| chan_iax2: Asterisk crashes when queueing video with format | (Reported by Michael Welk) | |
| STUN timeout is silently delaying calls | (Reported by Sébastien Duthil) | |
| Remote URL in playback must end with file extension | (Reported by Caesar) | |
| ari: Audiosocket segfault when no data specified | (Reported by Igor Goncharovsky) | |
| Updated identify/match syntax not supported by config wizard | (Reported by Sean Bright) | |
| fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew | (Reported by Dan Cropp) | |
| core: Inband generation of tones for Busy() and Congestion() may not occur | (Reported by Joshua C. Colp) | |
| [patch] Channels are not put on hold for Session Progress with inactive audio | (Reported by Bernd Zobl) | |
| 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) | |
| translate.c: possible buffer overflow when upsampling | (Reported by Jean Aunis - Prescom) | |
| Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 | (Reported by Ross Beer) | |
| prometheus: Crash when scraping bridge | (Reported by Francisco Correia) | |
| res_rtp_asterisk: standard deviation miscalculation | (Reported by Kevin Harwell) | |
| res_rtp_asterisk: Flash events are duplicated | (Reported by N A) | |
| app_queue: CLI set ringinuse for realtime member not working | (Reported by Michael) | |
| Fix differing usage of assignment operators in modules.conf | (Reported by Rusty Newton) | |
| Incorrect description of option "context" in queues.conf.sample | (Reported by Etienne Lessard) | |
| app_queue: updatecdr option in queues.conf does effectively nothing | (Reported by Alexander Gonchiy) | |
| dateformat not read from logger.conf by remote console | (Reported by Igor Liferenko) | |
| app_queue: When "queue show" CLI command is executed a crash occurs | (Reported by Miguel Sanz) | |
| res_pjsip_session: NULL active_media_state topology caused asterisk crash | (Reported by sungtae kim) | |
| app_queue: Queue member status message sent even if status doesn't change | (Reported by Roman Pertsev) | |
| chan_local: Multistream support breaks T.38 faxing | (Reported by Matthias Hensler) | |
| res_pjsip: Allow partial reloading of transports | (Reported by Joshua C. Colp) | |
| menuselect doesn't return errors in many cases | (Reported by George Joseph) | |
| res_rtp_asterisk: Fix frame delivery time when SSRC changes | (Reported by Joshua C. Colp) | |
| app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs | (Reported by Stefan Ruf) | |
| app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events | (Reported by N A) | |
| strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition | (Reported by Vitezslav Novy) | |
| res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent | (Reported by Sebastian Damm) | |
| res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address | (Reported by Brian Paboojian) | |
| ICE Role conflict with an unauthorized session | (Reported by Salah Ahmed) | |
| chan_pjsip: 180 Ringing with SDP not changed into progress | (Reported by Sebastian Damm) | |
| say: Y2021 problem â?? Asterisk cannot say year 2021 in Dutch | (Reported by Jacek Konieczny) | |
| res_pjsip: re-registration gets stuck if setting initial auth credentials fails | (Reported by Nick French) | |
| res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters | (Reported by Alexei Gradinari) | |
| Callee declined when 'beep' audio file does not exist | (Reported by IAMJames_) | |
| res_pjsip_refer: Segfault in progress notify | (Reported by George Joseph) | |
| res_config_pgsql: Limit realtime_pgsql() to return one (no more) record | (Reported by Boris P. Korzun) | |
| pjsip: Re-invite occurs when it shouldn't | (Reported by Benjamin Keith Ford) | |
| res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit | (Reported by Jaco Kroon) | |
| pjsip: | (Reported by Michael Maier) | |
| app.h: C++ compatibility broken | (Reported by Jean Aunis - Prescom) | |
| app_queue: Member device state "invalid" when second call is ringing and hint is used | (Reported by Boolah ) | |
| res_pjsip_t38: Crash when changing state | (Reported by Gregory Massel) | |
| res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client | (Reported by Edvin Vidmar) | |
| res_pjsip: Segmentation fault | (Reported by Mauri de Souza Meneguzzo (3CPlus)) | |
| chan_sip: Allow peers without audio (text+video). | (Reported by Alexander Traud) | |
| chan_sip: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| res_pjsip: user=phone validation fail for isup numbers containing *# | (Reported by Mark Petersen) | |
| channel: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| chan_sip: Audio stream rejected, Other stream present: Invalid SDP. | (Reported by Alexander Traud) | |
| After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used | (Reported by Robert Cripps) | |
| res_pjsip_session: res sometimes uninitialized reported by compiler Clang. | (Reported by Alexander Traud) | |
| Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription | (Reported by Jean Aunis - Prescom) | |
| chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable | (Reported by Ivan Poddubny) | |
| chan_sip: SDP: Offers without any enabled stream are accepted. | (Reported by Alexander Traud) | |
| chan_sip: SDP: m=video is parsed even when disabled. | (Reported by Alexander Traud) | |
| chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. | (Reported by Alexander Traud) | |
| chan_pjsip isn't updating hangupcause on 4XX responses | (Reported by George Joseph) | |
| PJSIP sends duplicate 183 Progress responses | (Reported by Alex Hermann) | |
| chan_pjsip: Subsequent same responses are not stopped | (Reported by Julien) | |
| pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send | (Reported by Michael Maier) | |
| pjsip: SIGSEGV in CLI if no trunk is registered | (Reported by Michael Maier) | |
| LOCK() can grant the same lock to multiple channels spuriously | (Reported by Jaco Kroon) | |
| Crash occurs when Transfer and execute Hangup before the Transfer result | (Reported by Dan Cropp) | |
| Segmentation fault in mixmonitor_ds_destroy | (Reported by Robert Sutton) | |
| Asterisk crashes during call transfer | (Reported by Dalius Mockevicius) | |
| res_pjsip: Crash when examining transport | (Reported by N GM ) | |
| tel: URI in Diversion header causes crash | (Reported by Mikhail Ivanov) | |
| Spyee information ist missing in ChanSpyStop AMI Event | (Reported by Hendrik Wedhorn) | |
| null media causing the Asterisk crash | (Reported by sungtae kim) | |
| pjsip: Route Header in Cancel request incorrectly set | (Reported by Flole Systems) | |
| Debug messages printed by scope trace might be missing newlines | (Reported by Alexander Traud) | |
| res_musiconhold: Segfault on realtime music on hold without entries | (Reported by Nathan Bruning) | |
| Crash when manipulating PJSIP invite dlg ref counts | (Reported by Sean Bright) | |
| Media cache URL requests allow infinite redirects | (Reported by Sean Bright) | |
| res_pjsip_stir_shaken: Fix module description | (Reported by Stanislav Abramenkov) | |
| AST_MODULE_INFO no, MODULEINFO depend | (Reported by Alexander Traud) | |
| res_pjsip: malformed header Accept-Encoding in OPTIONS response | (Reported by Alexander Greiner-Baer) | |
| [patch] chan_sip: TCP/TLS client without server. | (Reported by Alexander Traud) | |
| Incorrect setup of recall channels | (Reported by Boris P. Korzun) | |
| app_queue: Deadlock between queues container and individual queues | (Reported by George Joseph) | |
| res_pjsip.so fails to load when bundled pjproject is compiled without libssl | (Reported by Walter Doekes) | |
| Any curl response checks out as valid even if 404 is returned. | (Reported by dovid) | |
| res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies | (Reported by Sebastian Damm) | |
| sip_to_pjsip.py: doesn't read globbed includes | (Reported by Michael Newton) | |
| GCC Warnings with OPTIMIZE=-Og make | (Reported by Alexander Traud) | |
| GCC Warnings with OPTIMIZE=-Os make | (Reported by Alexander Traud) | |
| GCC Warnings: â??%sâ?? directive argument is null. | (Reported by Alexander Traud) | |
| res_pjsip: flow transport broken for outbound requests | (Reported by Nick French) | |
| config: Sample features.conf incorrectly includes " around sound files | (Reported by Benjamin M.) | |
| logger.conf.sample missing comment mark on line 115 | (Reported by Andrew Siplas) | |
| res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 | (Reported by Ross Beer) | |
| res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF | (Reported by under) | |
| resource_endpoints.c : Memory leak if endpoint not found | (Reported by Jean Aunis - Prescom) | |
| app_voicemail: Undocumented behavior from VMSayName | (Reported by Eric Smith) | |
| res_pjsip_config_wizard: Crash when freeing string when failing to add extension | (Reported by Vieri) | |
| Crash when ast_translator_build_path fails | (Reported by Jasper van der Neut) | |
| res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used | (Reported by Sebastian Damm) | |
| res_musiconhold: Realtime MOH only loads a single entry | (Reported by laszlovl) | |
| dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format | (Reported by �家建) | |
| Music On Hold announcement cuts intro of music the first time it is played | (Reported by Thomas Frederiksen) | |
| func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT | (Reported by Péter Juhász) | |
| RTP Ports not cleared after hangup | (Reported by Ross Beer) | |
| res_stasis: Add compare function for bridges moh container | (Reported by Hajek Michal) | |
| Unable to get rtp codec payload code for slin | (Reported by Brian J. Murrell) | |
| res_pjsip_session: Re-INVITE collisions aren't handled correctly | (Reported by George Joseph) | |
| Duplicate logging in queue log for EXITEMPTY events | (Reported by Ove Aursand) | |
| app_queue: Leave empty sometimes not recorded as abandoned | (Reported by Kfir Itzhak) | |
| res_parking: Parker UUID is no longer copied | (Reported by Misha Vodsedalek) | |
| chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 | (Reported by Joseph Ades) | |
| pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension | (Reported by Ramarajan) | |
| res_speech: Assertion on format | (Reported by Nickolay V. Shmyrev) | |
| chan_pjsip does not process or forward 181 responses | (Reported by Torrey Searle) | |
| Lastpause of realtime members is reseting | (Reported by Evandro César Arruda) | |
| app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command | (Reported by Leandro Dardini) | |
| res_pjsip_session: Aggressively terminates session on failed re-INVITE | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk: T.140 messages have appended RTP string to each message block. | (Reported by Thomas Johnson) | |
| chan_sip: ToHost property not cleared on reload | (Reported by Dennis) | |
| [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions | (Reported by cmaj) | |
| Asterisk crash in music on hold | (Reported by David Cunningham) | |
| Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) | (Reported by Michael Neuhauser) | |
| res_pjsip_registrar: Expires on statically configured contacts is not correct | (Reported by tootai) | |
| BridgeCreated ARI event shows wrong video_mode info | (Reported by sungtae kim) | |
| acl: named_acl rule misconfiguration results in segfault on reading rule from realtime | (Reported by Andrew Yager) | |
| res_http_websocket: Text payload data doesn't necessary include trailing zero | (Reported by Nickolay V. Shmyrev) | |
| Inconsistent behaviour queues.conf when there is (not) a [general] section | (Reported by Walter Doekes) | |
| res_pjsip: Apply outbound proxy to static contacts on AOR | (Reported by Joshua C. Colp) | |
| ./configure --without-ssl build failure | (Reported by Jaco Kroon) | |
| chan_sip: chan_sip does not process 400 response to an INVITE. | (Reported by Frederic LE FOLL) | |
| chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 | (Reported by Jared Smith) | |
| res_corosync: causes asterisk crash in huge distributed environment. | (Reported by Università di Bologna - CESIA VoIP) | |
| StreamEcho() only returns 1 active stream | (Reported by Bill Kervaski) | |
| "setvar" doesn't work properly in dahdi-channels.conf | (Reported by Marin Odrljin) | |
| res_pjsip_session: Preserve stream label | (Reported by Joshua C. Colp) | |
| res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching | (Reported by Joshua C. Colp) | |
| Stale code in app_queue to check untouched channel | (Reported by Walter Doekes) | |
| Stale comment in app_queue about ring_entry exception | (Reported by Walter Doekes) | |
| Queue wrapuptime sometimes not respected (based on stale lastcall time) | (Reported by Walter Doekes) | |
| core_unreal / core_local: Add support for multistream and re-negotiation | (Reported by Joshua C. Colp) | |
| ARI channel create doesn't referencing the channel_id parameter | (Reported by sungtae kim) | |
| res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC | (Reported by Joshua C. Colp) | |
| bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation | (Reported by Joshua C. Colp) | |
| T.38 Segfaults in chan_pjsip_queryoption | (Reported by Yury Kirsanov) | |
| /channels/create doesn't get any parameters from the body | (Reported by sungtae kim) | |
| res_pjsip: crash when dialing non-sip uri | (Reported by Walter Doekes) | |
| res_fax: Double frame free when gateway in use with off-nominal format usage | (Reported by Gregory Massel) | |
| pjproject_bundled: Honor --without-pjproject. | (Reported by Alexander Traud) | |
| res_pjsip_logger writing too big packets | (Reported by nappsoft) | |
| bridge show all causes crash | (Reported by sungtae kim) | |
| Wrong return value check for fwrite when writing to pcap file | (Reported by nappsoft) | |
| res_pjsip: Crash when escaping during URI printing | (Reported by nappsoft) | |
| x-ast-orig-host not filtered out from request URI and To header | (Reported by nappsoft) | |
| res_pjsip_session: Unnecessary re-Invite on call answer | (Reported by Alexei Gradinari) | |
| res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. | (Reported by Alexander Traud) | |
| bridge_softmix: Conference bridge not passing silent rtp packets | (Reported by Jonathan Hunter) | |
| res_musiconhold: Module res_musiconhold throws false warning | (Reported by Nicholas John Koch) | |
| RTP ICE leaks the memory | (Reported by sungtae kim) | |
| res_pjsip: PJSIP Registration Fails when transport=transport-udp6 | (Reported by Peter Sokolov) | |
| SIGSEGV when pjsip show history encounters IPV6 address | (Reported by Roger James) | |
| [patch] tcptls: Fix notice when TLS is enabled but not configured. | (Reported by Alexander Traud) | |
| [patch] app_osplookup.c: Avoid a format truncation. | (Reported by Alexander Traud) | |
| Non async-signal-safe syscalls used after fork before exec | (Reported by nappsoft) | |
| streams: One memory leak and one issue cloning streams | (Reported by George Joseph) | |
| app_queue: leaking stasis subscription when Redirecting call | (Reported by laszlovl) | |
| app_queue: Ghost channels in "core show channels" output | (Reported by Etienne Lessard) | |
| pjsip: Increase maximum candidate count | (Reported by Joshua C. Colp) | |
| Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling | (Reported by Shlomi Gutman) | |
| Unprotected access to nochecksums variable, causes build failures | (Reported by Guido Falsi) | |
| app_fax: Compile. | (Reported by Alexander Traud) | |
| stream: Enforce formats immutability | (Reported by Joshua C. Colp) | |
| ARI channels cuts the endpoint string over 80 characters | (Reported by sungtae kim) | |
| Crash occurs when fax session switches from T.38 to audio | (Reported by Alexey Vasilyev) | |
| Sporadic crashes with Segmentation fault | (Reported by Joeran Vinzens) | |
| IPv6 addresses in SDP incorrectly formatted | (Reported by Daniel Heckl) | |
| Asterisk REPLY Wrong Contact header port (TCP) | (Reported by Anton Satskiy) | |
| Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used | (Reported by sstream) | |
| AST_MODULE_INFO requires, MODULEINFO does not mention | (Reported by Alexander Traud) | |
| app_confbridge: Add support for disabling text messaging for a user | (Reported by Joshua C. Colp) | |
| pjproject_bundled: Honor --without-pjproject. | (Reported by Alexander Traud) | |
| res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK | (Reported by nappsoft) | |
| chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets | (Reported by Joshua Roys) | |
| res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK | (Reported by nappsoft) | |
| First DTMF is not get | (Reported by Bernard Merindol) | |
| pjsip startup errors when using "with-ssl" configure option | (Reported by Patrick Wakano) | |
| BuildSystem: Search for Python/C API when possibly needed only. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. | (Reported by Alexander Traud) | |
| chan_pjsip: constant DTMF tone if RTP is not setup yet | (Reported by Kevin Harwell) | |
| [patch] bridge_softmix_binaural: Show state in menuselect. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Allow space in path. | (Reported by Alexander Traud) | |
| [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. | (Reported by Alexander Traud) | |
| func_channel: cannot read fields exten, context, userfield, channame from dialplan | (Reported by Sébastien Duthil) | |
| [patch] chan_unistim: Avoid tautological warnings with clang. | (Reported by Alexander Traud) | |
| [patch] test_stasis: Avoid always true warning with clang. | (Reported by Alexander Traud) | |
| res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR | (Reported by Jason Hord) | |
| channel: write to a stream on multi-frame writes | (Reported by Kevin Harwell) | |
| test_utils: incorrectly printing error 'declined to load' | (Reported by Alexander Traud) | |
| func_aes: incorrectly printing error 'declined to load' | (Reported by Alexander Traud) | |
| Crash during conference call using confbridge and video | (Reported by Pascal Cadotte Michaud) | |
| DAHDIRAS fails to properly initiate pppd unless asterisk is running as root | (Reported by Jaco Kroon) | |
| [patch] dundi_read_result crash due to negative number | (Reported by Jaco Kroon) | |
| res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream | (Reported by Joshua C. Colp) | |
| Asterisk is crashing if the 200 OK with SDP | (Reported by sungtae kim) | |
| res_pjsip_session: Allow default non-audio streams to have reflected state | (Reported by Joshua C. Colp) | |
| chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge | (Reported by Michael Neuhauser) | |
| Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. | (Reported by Olivier Krief) | |
| app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup | (Reported by Joshua C. Colp) | |
| Incorrect Sender SSRC in RTCP when p2p rtp bridge is active | (Reported by Torrey Searle) | |
| DTLS Handshake Fails to Occur if ice_support is enabled but not used | (Reported by Torrey Searle) | |
| A non negotiated rtp frame causes call disconnection when there is a SSRC change | (Reported by Paulo Vicentini) | |
| func_enum: ENUM code wrong case | (Reported by Vitold) | |
| Fix the FSF address in the headers of lots of pjproject files | (Reported by Jared Smith) | |
| [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string | (Reported by George Joseph) | |
| PJSIP blind transfer not completed after using Proceeding() | (Reported by laszlovl) | |
| res_rtp_asterisk: Improve NACK support and seqno handling | (Reported by Joshua C. Colp) | |
| SIP/Stasis: SIP headers not transmitted in the "variables" field | (Reported by Jean Aunis - Prescom) | |
| check_expr2: linking (when hardening) and cross-compiling troubles | (Reported by Sebastian Kemper) | |
| ASTERISK-28738 Causes Audio Issue After Hold | (Reported by Ross Beer) | |
| res_pjsip: Named ACL does not update on reload if changed | (Reported by Timothy Vanderaerden) | |
| res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set | (Reported by George Joseph) | |
| ICE: pjnath shouldn't wait for ICE to complete before allowing sending | (Reported by Benjamin Keith Ford) | |
| Incorrect state machine used when MOH_PASSTHRU is used | (Reported by Torrey Searle) | |
| res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup | (Reported by Kevin Harwell) | |
| Realtime MoH Unknown format '' -- defaulting to SLIN | (Reported by Ross Beer) | |
| res_pjsip_session: Fix out of order session refreshes | (Reported by Joshua C. Colp) | |
| pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected | (Reported by Peter Sokolov) | |
| chan_sip: Returns 403 if RTP ports are depleted, should return 503 | (Reported by Walter Doekes) | |
| res_stasis_playback: Error building JSON | (Reported by Sébastien Duthil) | |
| REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults | (Reported by Ross Beer) | |
| res_pjsip_messaging: MessageSend Content-Type can't be changed | (Reported by Alex) | |
| ARI causes STASIS Deadlock | (Reported by Ross Beer) | |
| stasis application is destroyed after its creation | (Reported by Francois Blackburn) | |
| PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending | (Reported by Dmitriy Serov) | |
| chan_sip strictrtp=yes fails when media source is changed: no audio | (Reported by Walter Doekes) | |
| RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls | (Reported by Paul Brooks) | |
| CDR billsec is always 0 for transferred calls | (Reported by Maciej Michno) | |
| chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 | (Reported by Andrew Siplas) | |
| Update documentation for statsd module - usage requirements unclear | (Reported by Dan Jenkins) | |
| silk 24hHz doesn't show up in 'core show translation' output | (Reported by Sean Bright) | |
| core: minmemfree watermark uses free RAM, not available RAM | (Reported by Kevin Flyn) | |
| chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan | (Reported by Frank Matano) | |
| [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used | (Reported by Stas Kobzar) | |
| empty voicemail.conf required for ARA (realtime) voicemail to leave message | (Reported by Jim Van Meggelen) | |
| CLI command 'realtime update2' syntax failure when using according to usage help | (Reported by Cedric BASSAGET) | |
| Pause reason not reported in QueueMember AMI event | (Reported by Niksa Baldun) | |
| res_pjsip_endpoint_identifier_ip: Document support for hostnames | (Reported by Joshua C. Colp) | |
| res_pjsip_notify: Multiple Event headers can be present instead of just one | (Reported by AvayaXAsterisk) | |
| app_record: Lack of `beep` audio file causes application to return error and hangup | (Reported by Corey Farrell) | |
| Wiki docs missing for MessageWaiting | (Reported by David M. Lee) | |
| res_pjsip_pubsub: Subscription persistence does not preserve XML | (Reported by Bryan Nelson) | |
| chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X | (Reported by Dirk Wendland) | |
| stasis bridge topic leak | (Reported by Joeran Vinzens) | |
| pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group | (Reported by Jean-Denis Girard) | |
| SIP WSS message not processed until next frame arrives | (Reported by Robert Sutton) | |
| Asterisk ignores parsing of config files if a Byte order mark is present | (Reported by Robin Leffmann) | |
| Playback of local files impacted by large media cache | (Reported by Kevin Reeves) | |
| contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax | (Reported by Richard Kenner) | |
| "trustrpid" is misspelled in sip_to_pjsip.py | (Reported by Pascal Cadotte Michaud) | |
| app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. | (Reported by Frederic LE FOLL) | |
| app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 | (Reported by George Joseph) | |
| res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them | (Reported by nappsoft) | |
| res_fax: wrap Asterisk initiated negotiation with config option | (Reported by Kevin Harwell) | |
| Missing arguments in PJSIP_CONTACT function documentation | (Reported by Pascal Cadotte Michaud) | |
| Memory Leak in res_rtp_asterisk.c | (Reported by Ted G) | |
| chan_sip logs errors on tx to non-existent TCP connections | (Reported by Jaco Kroon) | |
| chan_pjsip incorrectly re-writes REGISTER 200 Response Contact | (Reported by Ross Beer) | |
| res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR | (Reported by Ross Beer) | |
| chan_sip: RTP frames not transmitted after emitting a COLP | (Reported by Jean Aunis - Prescom) | |
| chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. | (Reported by Frederic LE FOLL) | |
| res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled | (Reported by Bernhard Schmidt) | |
| res_parking: Doesn't park when parkee and parker are the same | (Reported by Ross Beer) | |
| Enforce T.38 error correction mode at 200 ok received | (Reported by Salah Ahmed) | |
| res_pjsip_outbound_registration: add SRV failover | (Reported by Kevin Harwell) | |
| app_amd: Use time calculation to calculate timeout | (Reported by Michael Cargile) | |
| chan_dahdi: PRI span status may stay "Down, Active" after a short alarm | (Reported by Frederic LE FOLL) | |
| res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match | (Reported by Joshua Elson) | |
| FILE function grabs garbage along with read data when target line has no newline | (Reported by Jonathan Harris) | |
| bridge_softmix: hold not cleared when joining a softmix bridge | (Reported by Kevin Harwell) | |
| parking: Deadlock when multi call parking | (Reported by Joshua C. Colp) | |
| Memory leaks in res_calendar_exchange and res_calendar_icalendar | (Reported by Yoooooo Ha) | |
| ari/resource_events: Crash in event session cleanup | (Reported by Kevin Harwell) | |
| utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" | (Reported by Speed Dial Dave) | |
| race condition on pjsip channelstats command | (Reported by Salah Ahmed) | |
| cdr_pgsql: accesses obsolete (and finally removed) column | (Reported by Christoph Moench-Tegeder) | |
| MWI Send Notify Crash on 16.6 | (Reported by Joshua Elson) | |
| pjproject fails to build on 16.6.0, works on 16.5 | (Reported by Niklas Larsson) | |
| Asterisk Deadlocks | (Reported by Aheliotech) | |
| chan_pjsip: Crash when initiating PlayDTMF over AMI | (Reported by Jeremiah Gadd) | |
| res_pjsip_mwi: Frack during unload on unsolicited_mwi container | (Reported by Kevin Harwell) | |
| CDR backend unload problem during active call(s) | (Reported by Marian Piater) | |
| stasis.c: Crash during unload | (Reported by Kevin Harwell) | |
| Wrong contact representation in ipv6 mode | (Reported by Jørgen H) | |
| Segmentation fault when there is no priority for an extension | (Reported by Timothy Vanderaerden) | |
| res_pjsip_path: Crash when invalid contact is configured | (Reported by Juan Martin) | |
| pjsip: Memory Leak | (Reported by Mark) | |
| Asterisk 16.5.0 Memory leak | (Reported by Cyril Ramière) | |
| Asterisk release candidates fail to build on FreeBSD | (Reported by Guido Falsi) | |
| chan_pjsip: Deadlock on fax detection | (Reported by Joshua C. Colp) | |
| func_odbc: truncating Unicode string on readsql | (Reported by Boris P. Korzun) | |
| setvar directive when used in template and a child of said template, results in duplicate variable names | (Reported by Michael Goryainov) | |
| ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf | (Reported by Frederic LE FOLL) | |
| chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up | (Reported by Frederic LE FOLL) | |
| codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 | (Reported by Ruddy G) | |
| translate: Crash when frame does not have a "src" field set | (Reported by Gregory Massel) | |
| chan_unistim: Clang Warning: variable sized type not at end of a struct | (Reported by Alexander Traud) | |
| pjsip mwi: n+1 sip notify's sent on re-register | (Reported by Chris Savinovich) | |
| PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters | (Reported by Dan Cropp) | |
| app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream | (Reported by Alexei Gradinari) | |
| compile menuselect on gentoo | (Reported by Kilburn) | |
| Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV | (Reported by Jonas Swiatek) | |
| cel / cdr: Event times may be incorrect | (Reported by Joshua C. Colp) | |
| json integer overflow in ssrc and timestamp | (Reported by Salah Ahmed) | |
| res_pjsip: pjsip show contacts prints double entries | (Reported by Ian Jones) | |
| packet lost on UDPTL wrap around | (Reported by Torrey Searle) | |
| Crash when not specifying "dbfile" in res_config_sqlite3.conf | (Reported by Dennis) | |
| Crash performing "core reload" with modified res_config_sqlite3.conf | (Reported by Dennis) | |
| AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) | (Reported by Walter Doekes) | |
| res_pjsip_mwi: Memory leak on reload | (Reported by Sergej Kasumovic) | |
| [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 | (Reported by abelbeck) | |
| res_pjsip_sdp_rtp: Remove unused variable | (Reported by Michael Maier) | |
| Show offending IP for TLS setup failures in logs | (Reported by Oleksandr Natalenko) | |
| chan_pjsip: Peer IP for SSL handshake errors not logged | (Reported by Bernhard Schmidt) | |
| chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer | (Reported by Dan Cropp) | |
| app_amd: Does not work with silence suppression | (Reported by Nasir Iqbal) | |
| IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate | (Reported by vijay kumar) | |
| Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event | (Reported by Abhay Gupta) | |
| cdr_pgsql: Unix socket doesn't work | (Reported by Dmitry Svyatogorov) | |
| res_fax: Fax session leak with fax gatewaying | (Reported by pasandev) | |
| new mwi.h include missing from some dahdi source files, causes build failure | (Reported by Guido Falsi) | |
| Wrong type used for timestamp in res_rtp_asterisk | (Reported by Morten Tryfoss) | |
| Removal of Previous Patch Causes PJSIP Timer Issues | (Reported by Ross Beer) | |
| PJSIP: Early media ringback not indicated after Progress() | (Reported by Gregory Massel) | |
| GCC 9 catches more string formatting issues | (Reported by George Joseph) | |
| pjsip: show channelstats incorrect information output | (Reported by Vyrva Igor) | |
| channel.c: Exceptionally long queue length queuing | (Reported by Abhay Gupta) | |
| The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds | (Reported by George Joseph) | |
| res_pjsip_registrar: SEGV in registrar_find_contact | (Reported by Ross Beer) | |
| bridge: Failure to impart a channel results in bad data causing crash | (Reported by Abhay Gupta) | |
| ARI: Bridge destroying doesn't work as expected | (Reported by Marin Odrljin) | |
| app_amd: Infinite loop on silent calls | (Reported by Abhay Gupta) | |
| stasis: Crash at shutdown when statistics enabled | (Reported by Joshua C. Colp) | |
| latest asterisk unconditionally launch gcc --version, even if the compiler is different | (Reported by Guido Falsi) | |
| res_indications: Crash requesting autocomplete on indications cli command | (Reported by Lucas Mendes) | |
| app_voicemail: emailbody per user can't contain commas | (Reported by Sébastien Duthil) | |
| 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them | (Reported by test011) | |
| AEL reload causes loss of control in a macro | (Reported by Kirill Katsnelson) | |
| AEL for loops use Macro app and pipe delimiter | (Reported by Luke-Jr) | |
| AEL parsers does not find existing label | (Reported by klaus3000) | |
| Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIfTime application causes unexpected behavior | (Reported by Janu) | |
| Failed to initialize OOH323 endpoint-OOH323 Disabled | (Reported by Dmitry Shubin) | |
| chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info | (Reported by Salah Ahmed) | |
| musl: Crash on startup when loading modules | (Reported by Sebastian Kemper) | |
| strtok_r() makes gcc compile warning | (Reported by sungtae kim) | |
| res_rtp_asterisk: REMB RTCP packet sending may be incorrect | (Reported by Joshua C. Colp) | |
| app_queue: Queue paused reason was (big number) secs ago when reason is set | (Reported by César BenjamÃn GarcÃa MartÃnez) | |
| QUEUE_MEMBER 's description is inaccurate | (Reported by Olivier Krief) | |
| manager: Stasis backed up due to locking | (Reported by Joshua C. Colp) | |
| chan_sip: qualifygap bounds checking | (Reported by Paul Sandys) | |
| res_config_odbc eliminates empty custom (â??@â?? prefix) variables | (Reported by Alexei Gradinari) | |
| StasisEnd event makes wrong timestamp value | (Reported by sungtae kim) | |
| res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent | (Reported by Jared Hull) | |
| Variable ALTCONF ignored when service is used in Debian | (Reported by Cirillo Ferreira) | |
| app_queue: ring_entry accesses nativeformats without channel lock or reference | (Reported by Francisco Seratti) | |
| stasis: Make topic and maybe subscription names unique and more useful | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation | (Reported by sungtae kim) | |
| chan_pjsip: Add option to allow ignoring of 183 without SDP | (Reported by Torrey Searle) | |
| MeetMe global non-admin mute is muting admins that subsequently join | (Reported by Philip Mott) | |
| app_queue: Adding a blank entry into sql queue_members crashes asterisk. | (Reported by Michael) | |
| pjsip: sip.conf to pjsip.conf conversion script fails | (Reported by Guido Weckwerth) | |
| The basic-pbx config samples don't produce a running asterisk | (Reported by George Joseph) | |
| res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect | (Reported by Alex Odrov) | |
| File menuselect/menuselect_gtk.c has no license header | (Reported by Jeremy Lainé) | |
| app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC | (Reported by Michael) | |
| res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces | (Reported by Nikolay shakin) | |
| PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash | (Reported by Jonathan Harris) | |
| res_pjsip: Threads pile up needlessly when AOR is blocked | (Reported by Ross Beer) | |
| Allow voicemail boxes to be subscribed to with a presence event package | (Reported by George Joseph) | |
| res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps | (Reported by Torrey Searle) | |
| ARI: "Error destroying mutex" when listing all ARI applications | (Reported by Stefan Repke) | |
| AST_PBX_MAX_STACK is too low for some applications | (Reported by George Joseph) | |
| Astricon Feedback: Unable to filter ARI events when GETting causes overload of events | (Reported by George Joseph) | |
| switching between native_bridge and simple_bridge can cause one way audio | (Reported by Torrey Searle) | |
| CI: Fix CI so it reverifies commit message changes | (Reported by George Joseph) | |
| database: Add some basic logging | (Reported by Joshua C. Colp) | |
| ari: Originating overwrites channel start time | (Reported by sungtae kim) | |
| Deadlock in chan_sip handling subscribe request during res_parking reload | (Reported by Giuseppe Sucameli) | |
| AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps | (Reported by George Joseph) | |
| Opensuse Leap 15 --with-jannson-bundled will not compile | (Reported by David Wilcox) | |
| PJSIP realtime. getcontext not working with DUNDI | (Reported by Ray) | |
| codec_opus: errors setting max_playback_rate and bitrate to "sdp" | (Reported by Gianluca Merlo) | |
| res_http_websocket: PING / PONG opcodes break data reception | (Reported by Jeremy Lainé) | |
| build: Cross-compilation fails for target arm-linux-gnueabihf | (Reported by Jean Aunis - Prescom) | |
| HangupHandler manager events are never thrown | (Reported by Gerald Schnabel) | |
| res_http_websocket: Not responding to Connection Close Frame (opcode 8) | (Reported by Jeremy Lainé) | |
| res_monitor: Segfault with Monitor(wav,file,i) | (Reported by Valentin VidiÄ?) | |
| stasis: Filter messages at publishing to AMI/ARI | (Reported by Joshua C. Colp) | |
| stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases | (Reported by Mohit Dhiman) | |
| res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony | (Reported by David Kuehling) | |
| core: RAII using clang use-after-scope issue | (Reported by Diederik de Groot) | |
| [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation | (Reported by Alexei Gradinari) | |
| app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent" | (Reported by boatright) | |
| app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b) | (Reported by Mark) | |
| stasis: Statistics broke ABI under developer mode | (Reported by Joshua C. Colp) | |
| Regression: MWI polling no longer works | (Reported by abelbeck) | |
| Bug in ast_coredumper | (Reported by Andrew Nagy) | |
| app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs | (Reported by George Joseph) | |
| [patch] Asterisk 15.4.1 h264 fmtp negotiation problem | (Reported by David Kuehling) | |
| [patch] confbridge: no announce to the marked users when they join an empty conference | (Reported by Alexei Gradinari) | |
| stasis: Add statistics for usage when in developer mode | (Reported by Joshua C. Colp) | |
| stasis: Filter messages at publishing based on to_* presence | (Reported by Joshua C. Colp) | |
| chan_sip: Leak using contact ACL | (Reported by Giuseppe Sucameli) | |
| Asterisk crashes when the res_pjsip_* modules unload | (Reported by sungtae kim) | |
| app_queue: Revert broken queue channel reference patch | (Reported by laszlovl) | |
| chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE | (Reported by George Joseph) | |
| chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE | (Reported by nappsoft) | |
| app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default | (Reported by Ronald Raikes) | |
| stasis: Segment channel snapshot to reduce creation cost | (Reported by Joshua C. Colp) | |
| stasis: Use implementation specific cache for channel snapshots | (Reported by Joshua C. Colp) | |
| SIGABRT caused by stack corruption in hashkeys_read when no matching keys present | (Reported by Michael Walton) | |
| repeated segmentation faults | (Reported by Eyal Hasson) | |
| stasis: Filter messages at publishing to reduce work done | (Reported by Joshua C. Colp) | |
| ARI /channels/create handler causes core dump | (Reported by sungtae kim) | |
| Incorrect Behavior for rewrite_contact when Re-Invite omits routset | (Reported by Torrey Searle) | |
| Some conditions prevent running of el_end, break the terminal. | (Reported by Corey Farrell) | |
| rtp: Incorrect Packetization | (Reported by Robert Cripps) | |
| pbx_config: Only the first [globals] section is processed. | (Reported by Corey Farrell) | |
| Formatting error in documentation | (Reported by Scott Griepentrog) | |
| chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces | (Reported by Luit van Drongelen) | |
| res_pjsip_notify: improve realtime performance on CLI completion on the endpoint | (Reported by Alexei Gradinari) | |
| Caller ID cannot be changed on Attended Transfer before dialing out | (Reported by Alexei Gradinari) | |
| app_confbridge: Participant info labels aren't being added to the SDPs | (Reported by George Joseph) | |
| function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload | (Reported by Emmanuel BUU) | |
| bridging: Asterisk crashes when receiving an empty realtime text frame | (Reported by Emmanuel BUU) | |
| app_queue: QueueMemberStatus Event flooding AMI | (Reported by Andrej) | |
| res_pjsip: improve realtime performance on CLI 'pjsip show contacts' | (Reported by Alexei Gradinari) | |
| app_queue: Queue member considered inuse after immediately hanging up during dialing. | (Reported by Cao Minh Hiep) | |
| stasis: Playing MOH to bridge with ARI does not work | (Reported by Cameron) | |
| res_odbc: missing SQL error diagnostic | (Reported by Alexei Gradinari) | |
| chan_sip: SipNotify via AMI behaves differently to CLI | (Reported by Peter Katzmann) | |
| configure script does not enforce libunbound2 version | (Reported by Samuel Galarneau) | |
| testsuite: Sniffer assumes pjmedia will use ports below 10000 | (Reported by Joshua C. Colp) | |
| rtp: Crash in off-nominal case where RTP instance can't be set up | (Reported by Lei Fu) | |
| chan_sip unstable with TLS after asterisk start or reloads | (Reported by David Hajek) | |
| PJSIP: Update bundled PJPROJECT to version 2.8 | (Reported by Joshua C. Colp) | |
| chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs | (Reported by Will) | |
| AMI event "NewExten" is set to the wrong class | (Reported by laszlovl) | |
| res_pjproject build failure | (Reported by Jaco Kroon) | |
| [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file | (Reported by Frederic LE FOLL) | |
| channel.c: ARI ring only once | (Reported by Hajek Michal) | |
| Realtime queuemembers are not updated during retry phase | (Reported by laszlovl) | |
| alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean | (Reported by Joshua C. Colp) | |
| res_pjsip_transport_websocket: Properly set 'received' for IPv6 | (Reported by Sean Bright) | |
| When T.140 realtime text is negociated, a lot of debug traces are generated | (Reported by Emmanuel BUU) | |
| PBX calls via chan_sip TCP trunk now get authentification error | (Reported by Ian Gilmour) | |
| res_pjsip realtime: uri column in ps_contacts table can be too short | (Reported by Florian Floimair) | |
| res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE | (Reported by Joshua Elson) | |
| rtcp-mux is put in SDP answer regardless of offer | (Reported by Torrey Searle) | |
| No joint capabilities with video and audio-only streams | (Reported by Benjamin Keith Ford) | |
| app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY | (Reported by Valentin Safonov) | |
| pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. | (Reported by Alexander Traud) | |
| Wrong SRTP use status report | (Reported by Salah Ahmed) | |
| res_pjsip_registrar: Improve performance of inbound handling | (Reported by Joshua C. Colp) | |
| pjsip: Race condition in 183 re transmission can result in a deadlock | (Reported by Torrey Searle) | |
| make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o | (Reported by Majdi Bsoul) | |
| [regression] menuselect compilation failure on Solaris 10 | (Reported by Samuel Owens) | |
| menuselect compilation failure on Solaris 10 / gcc 3.4.3 | (Reported by rleasure) | |
| menuselect compilation failure on Solaris 10/gcc-4.1.1 | (Reported by Bob Atkins) | |
| BuildSystem: Enable Jansson in Solaris 11. | (Reported by Alexander Traud) | |
| res_pjsip_endpoint_identifier_ip only matches against "generic string" headers | (Reported by George Joseph) | |
| res_rtp_asterisk: Requires OpenSSL in Developer Mode. | (Reported by Alexander Traud) | |
| Frack errors in stasis.c and memory leakage | (Reported by Siruja Maharjan) | |
| res_pjsip: Change default transport keepalive to preserve behavior | (Reported by Joshua C. Colp) | |
| systemd: asterisk.service | (Reported by seanchann.zhou) |
Improvements made in this release:
-----------------------------------
| documentation: Standardize example syntax | (Reported by N A) | |
| app_voicemail: Refactor email generation functions | (Reported by N A) | |
| Add type for JSON stasis message RTCP Report Received/Sent | (Reported by Boris P. Korzun) | |
| Spelling errors | (Reported by Josh Soref) | |
| chan_iax2: Allow both key and secret to be specified at dial time | (Reported by N A) | |
| Add mix option to Playback application for say and filename | (Reported by Shloime Rosenblum) | |
| Add support for future dates in Say.c | (Reported by Shloime Rosenblum) | |
| PJSIP remove_existing unavailable contacts | (Reported by Joseph Nadiv) | |
| func_vmcount: Add support for multiple mailboxes | (Reported by N A) | |
| Support of MIME-type for wav16 | (Reported by Boris P. Korzun) | |
| Add custom logging level | (Reported by N A) | |
| res_pjsip: OLI/ANI2 support missing | (Reported by N A) | |
| app_stack: Include calling location if attempting to branch to nonexistent location | (Reported by N A) | |
| Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present | (Reported by Charlie Smurthwaite) | |
| chan_iax2: Add ANI2 | (Reported by N A) | |
| STUN server address refresh | (Reported by Sébastien Duthil) | |
| bridge_basic: Don't throw warning if attended transfer is cancelled | (Reported by N A) | |
| Media Cache - Delayed remote sound file retrieve delays all playbacks | (Reported by Andre Barbosa) | |
| Return integer instead of float if response is a whole number | (Reported by N A) | |
| app_morsecode: Add American Morse code | (Reported by N A) | |
| app_originate: Allow specifying codec(s) to use | (Reported by N A) | |
| Add support for multiple files for agent announcements | (Reported by N A) | |
| ARI - Stasis Playback doesn't hangup call when processing a list of invalid files | (Reported by Andre Barbosa) | |
| ARI - PlaybackFinish skip error events | (Reported by Andre Barbosa) | |
| 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) | |
| loader: Let's output warnings for deprecated modules! | (Reported by Joshua C. Colp) | |
| menuselect: Add ability to set deprecated in and removed in versions for modules | (Reported by Joshua C. Colp) | |
| documentation: Fix inconsistent support levels | (Reported by Joshua C. Colp) | |
| xml: Embed module information into core XML documentation. | (Reported by Joshua C. Colp) | |
| sorcery: Add support for more intelligent reloading. | (Reported by Joshua C. Colp) | |
| res_pjsip_registrar: Include source IP address and port in log messages | (Reported by Joshua C. Colp) | |
| asterisk: Update copyright/company | (Reported by Joshua C. Colp) | |
| Add MixMonitorStart / Stop / Mute AMI events | (Reported by Sébastien Duthil) | |
| TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code | (Reported by Dan Cropp) | |
| Support of various URL-schemes by MoH | (Reported by Boris P. Korzun) | |
| Two repeated 183 | (Reported by Gant Liu) | |
| contrib: systemd asterisk service for centos8 or other newer linux versions | (Reported by Mark Petersen) | |
| res_http_media_cache: HTTP media cache stored hardcoded in /tmp | (Reported by laszlovl) | |
| VoiceMail() should have an option to play greetings as Early Media | (Reported by Juan Carlos Castro y Castro) | |
| Logger: Add debug logging categories | (Reported by Kevin Harwell) | |
| Increase reg_server column size for ps_contacts table realtime | (Reported by sungtae kim) | |
| Create a Bridge with video_single mode | (Reported by sungtae kim) | |
| res_pjsip: Added option for disable rport parameter set | (Reported by sungtae kim) | |
| Continue reading string when ping received by websocket | (Reported by Nickolay V. Shmyrev) | |
| AMI SendText - add Content-Type parameter | (Reported by Kevin Harwell) | |
| res_http_websocket: Add masking to websocket client | (Reported by Moises Silva) | |
| Upgrade Asterisk to bundled pjproject 2.10 | (Reported by Kevin Harwell) | |
| res_pjsip_logger: Add tons'o'functionality | (Reported by Joshua C. Colp) | |
| ari: Add support for specifying variables on channel create | (Reported by Joshua C. Colp) | |
| pjproject has race conditions in it's build system | (Reported by Guido Falsi) | |
| third-party/pjproject/configure.m4 contains bashisms | (Reported by Guido Falsi) | |
| Missing include on FreeBSD | (Reported by Guido Falsi) | |
| chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio | (Reported by Peter Turczak) | |
| func_volume: Allow decimal numbers as parameter to improve granularity | (Reported by Jean Aunis - Prescom) | |
| Codec Negotiation: add outgoing_call_offer_prefs option | (Reported by Kevin Harwell) | |
| dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't | (Reported by Joshua Elson) | |
| Add support for Content-Disposition header in multi-part INVITES | (Reported by Torrey Searle) | |
| res_pjsip_session: Decide more intelligently when to add video | (Reported by Joshua C. Colp) | |
| Codec Negotiation: add incoming_call_offer_pref option | (Reported by Kevin Harwell) | |
| TLS/SSL Key too small error | (Reported by Martin Zeh) | |
| stream: Add support for adding/removing streams during SFU/calls | (Reported by Joshua C. Colp) | |
| Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor | (Reported by xrobau) | |
| install_prereq script uses the interactive mode when installing aptitude | (Reported by Sylvain Afchain) | |
| Should be able to disable the /httpstatus URI in the built-in HTTP server | (Reported by Sean Bright) | |
| Add AudioSocket support | (Reported by Seán C. McCord) | |
| Simplify dialplan for Dial, Page, and ChanIsAvail | (Reported by cmaj) | |
| GET FULL VARIABLE documentation clarification | (Reported by Jonathan Harris) | |
| [patch] Add an "inhibitCOLP" flag to the bridges REST API | (Reported by Jean Aunis - Prescom) | |
| app_confbridge: Add support for setting maximum sample rate | (Reported by Joshua C. Colp) | |
| res_pjsip_outbound_registration: Maximum retries reached | (Reported by Daniel) | |
| Typo in README-SERIOUSLY.bestpractices.md | (Reported by Sam Banks) | |
| [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column | (Reported by cmaj) | |
| Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. | (Reported by Michael) | |
| [patch] add the ability for asterisk to generate on-hold re-invites | (Reported by Torrey Searle) | |
| Add pass-through support for H.265 (HEVC) codec | (Reported by Florian Floimair) | |
| app_voicemail: remove dependency on stasis cache | (Reported by Kevin Harwell) | |
| stasis_state: Create a stasis module to cache last known state | (Reported by Kevin Harwell) | |
| res_ari_channels: Added detail hangup code settings | (Reported by sungtae kim) | |
| pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi | (Reported by Kirsty Tyerman) | |
| app_confbridge: Add *_all remb behavior variants | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc | (Reported by Joshua C. Colp) | |
| Millisecond-resolution call stats including PDD in channel variables | (Reported by Antoni Goldstein) | |
| Added detail subscriber/subscription info for stasis show app cli | (Reported by sungtae kim) | |
| Asterisk should clear out any .lock files in the voice mail directory on startup. | (Reported by Steven Wheeler) | |
| build: CHANGES/UPGRADE are irritating to work with. | (Reported by Corey Farrell) | |
| Added topic_all container | (Reported by sungtae kim) | |
| Added app_name, app_data to channel type | (Reported by sungtae kim) | |
| ari: Added timestamp for some ari events. | (Reported by sungtae kim) | |
| Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function | (Reported by Cirillo Ferreira) | |
| Added creation timestamp for bridge | (Reported by sungtae kim) | |
| Allow wrapuptime to be set for each queue member | (Reported by Rodrigo Ramirez Norambuena) | |
| app_queue: Per-member wrapup time missing from AddQueueMember application | (Reported by Niksa Baldun) | |
| Changed to show all channel stats including wrong media | (Reported by sungtae kim) | |
| res_pjsip_session: Adding rtcp stats result into the session | (Reported by sungtae kim) | |
| Support skipping on the g726 format | (Reported by Eyal Hasson) | |
| bridge_softmix: Does not support WebRTC source with multi video tracks. | (Reported by Xiemin Chen) | |
| res_ari: Add new hangup causes for ARI Channel DELETE command | (Reported by Sebastian Damm) | |
| [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI | (Reported by Alexei Gradinari) | |
| Allow the sip_to_pjsip script to be used in a pipe | (Reported by Pascal Cadotte Michaud) | |
| Remove stale nonoptreq references | (Reported by Walter Doekes) | |
| [patch] Add IPv6 Support for DUNDi | (Reported by Adam Secombe) | |
| PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID | (Reported by Eric Dantie) | |
| pjproject_bundled: Find shared libraries in root --with-ssl=PATH. | (Reported by Alexander Traud) | |
| pjsip_wizard example gives wrong info about unsupported SRV records | (Reported by Jonathan Harris) | |
| res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break | (Reported by Emmanuel BUU) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-18.9-cert1
Thank you for your continued support of Asterisk!
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