The release of Certified Asterisk 16.8-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:
Security bugs fixed in this release:
-----------------------------------
| 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) |
New Features made in this release:
-----------------------------------
| 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 lvl) | |
| 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) | |
| 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) |
Bugs fixed in this release:
-----------------------------------
| res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK | (Reported by nappsoft) | |
| res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK | (Reported by nappsoft) | |
| channel: write to a stream on multi-frame writes | (Reported by Kevin Harwell) | |
| Crash during conference call using confbridge and video | (Reported by Pascal Cadotte Michaud) | |
| res_pjsip_session: Allow default non-audio streams to have reflected state | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk: Improve NACK support and seqno handling | (Reported by Joshua C. Colp) | |
| res_pjsip_session: Fix out of order session refreshes | (Reported by Joshua C. Colp) | |
| res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set | (Reported by George Joseph) | |
| res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup | (Reported by Kevin Harwell) | |
| stasis application is destroyed after its creation | (Reported by Francois Blackburn) | |
| ARI causes STASIS Deadlock | (Reported by Ross Beer) | |
| REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults | (Reported by Ross Beer) | |
| 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) | |
| silk 24hHz doesn't show up in 'core show translation' output | (Reported by Sean Bright) | |
| Update documentation for statsd module - usage requirements unclear | (Reported by Dan Jenkins) | |
| 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) | |
| Pause reason not reported in QueueMember AMI event | (Reported by Niksa Baldun) | |
| CLI command 'realtime update2' syntax failure when using according to usage help | (Reported by Cedric BASSAGET) | |
| 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) | |
| contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax | (Reported by Richard Kenner) | |
| func_odbc: truncating Unicode string on readsql | (Reported by Boris P. Korzun) | |
| chan_sip: RTP frames not transmitted after emitting a COLP | (Reported by Jean Aunis - Prescom) | |
| Asterisk ignores parsing of config files if a Byte order mark is present | (Reported by Robin Leffmann) | |
| "trustrpid" is misspelled in sip_to_pjsip.py | (Reported by Pascal Cadotte Michaud) | |
| 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) | |
| app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. | (Reported by Frederic LE FOLL) | |
| 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) | |
| Stale comment in app_queue about ring_entry exception | (Reported by Walter Doekes) | |
| res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled | (Reported by Bernhard Schmidt) | |
| chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. | (Reported by Frederic LE FOLL) | |
| 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) | |
| Playback of local files impacted by large media cache | (Reported by Kevin Reeves) | |
| 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) | |
| 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) | |
| chan_pjsip: Crash when initiating PlayDTMF over AMI | (Reported by Jeremiah Gadd) | |
| 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) | |
| chan_pjsip: Deadlock on fax detection | (Reported by Joshua C. Colp) | |
| Asterisk release candidates fail to build on FreeBSD | (Reported by Guido Falsi) | |
| setvar directive when used in template and a child of said template, results in duplicate variable names | (Reported by Michael Goryainov) | |
| codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 | (Reported by Ruddy G) | |
| chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up | (Reported by Frederic LE FOLL) | |
| ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf | (Reported by Frederic LE FOLL) | |
| 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) | |
| chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer | (Reported by Dan Cropp) | |
| AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) | (Reported by Walter Doekes) | |
| [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) | |
| 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) | |
| 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) |
Improvements made in this release:
-----------------------------------
| res_pjsip_session: Decide more intelligently when to add video | (Reported by Joshua C. Colp) | |
| stream: Add support for adding/removing streams during SFU/calls | (Reported by Joshua C. Colp) | |
| Should be able to disable the /httpstatus URI in the built-in HTTP server | (Reported by Sean Bright) | |
| Simplify dialplan for Dial, Page, and ChanIsAvail | (Reported by cmaj) | |
| GET FULL VARIABLE documentation clarification | (Reported by Jonathan Harris) | |
| 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) | |
| 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) | |
| 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 app_name, app_data to channel type | (Reported by sungtae kim) | |
| Added topic_all container | (Reported by sungtae kim) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cert1
Thank you for your continued support of Asterisk!
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