[asterisk-dev] Asterisk 18.0.0 Now Available
John Kiniston
johnkiniston at gmail.com
Tue Oct 27 16:13:53 CDT 2020
Is anyone else seeing menuconfig give the wrong description app_audiosocket
and chan_audiosocket selections with this release?
I've tried on two systems and I'm seeing the same thing, If I highlight
app_audioosocket I get a description of AST_MODULE_INFO(
and chan_audiosocket has a description of AST_MODULE_INFO(ASTERISK_GPL_KEY,
AST_MODFLAG_LOAD_ORDER,
It's not affecting me, Just a weird display thing.
On Tue, Oct 20, 2020 at 5:02 AM Asterisk Development Team <
asteriskteam at digium.com> wrote:
> The Asterisk Development Team would like to announce the release of
> Asterisk 18.0.0.
> This release is available for immediate download at
> https://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 18.0.0 resolves several issues reported by the
> community and would have not been possible without your participation.
>
> *Thank you!*
>
> The following issues are resolved in this release:
>
> *Security bugs fixed in this release:*
> -----------------------------------
>
> - [ASTERISK-28589
> <https://issues.asterisk.org/jira/browse/ASTERISK-28589>] -
>
> chan_sip: Depending on configuration an INVITE can alter Addr of a peer
> (Reported by Andrey V. T.)
>
> - [ASTERISK-28580
> <https://issues.asterisk.org/jira/browse/ASTERISK-28580>] -
>
> Bypass SYSTEM write permission in manager action allows system commands
> execution
> (Reported by Eliel Sardañons)
>
> - [ASTERISK-28495
> <https://issues.asterisk.org/jira/browse/ASTERISK-28495>] -
>
> res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
> (Reported by Alexei Gradinari)
>
> *New Features made in this release:*
> -----------------------------------
>
> - [ASTERISK-6863
> <https://issues.asterisk.org/jira/browse/ASTERISK-6863>] -
>
> [patch] allow Asterisk to set high ToS bits as non-root on Linux
> (Reported by Matt Addison)
>
> - [ASTERISK-17491
> <https://issues.asterisk.org/jira/browse/ASTERISK-17491>] -
>
> CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do
> anything
> (Reported by candrews)
>
> - [ASTERISK-28639
> <https://issues.asterisk.org/jira/browse/ASTERISK-28639>] -
>
> res_pjsip_endpoint_identifier_ip: Add ability to match on source port
> (Reported by Sean Bright)
>
> - [ASTERISK-28614
> <https://issues.asterisk.org/jira/browse/ASTERISK-28614>] -
>
> app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only
> "sending"
> (Reported by lvl)
>
> - [ASTERISK-28613
> <https://issues.asterisk.org/jira/browse/ASTERISK-28613>] -
>
> func_curl: CURLOPT cannot set Content-Type header
> (Reported by Martin Tomec)
>
> - [ASTERISK-28533
> <https://issues.asterisk.org/jira/browse/ASTERISK-28533>] -
>
> func_jitterbuffer: Add support for video synchronization
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-17808
> <https://issues.asterisk.org/jira/browse/ASTERISK-17808>] -
>
> [patch] Unregister a realtime moh class
> (Reported by Byron Clark)
>
> - [ASTERISK-28489
> <https://issues.asterisk.org/jira/browse/ASTERISK-28489>] -
>
> Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI
> domain
> (Reported by Stas Kobzar)
>
> *Bugs fixed in this release:*
> -----------------------------------
>
> - [ASTERISK-29109
> <https://issues.asterisk.org/jira/browse/ASTERISK-29109>] -
>
> res_pjsip_session: Asterisk 18 does not progress calls due to codec
> negotiation after upgrading from Asterisk 16
> (Reported by Ross Beer)
>
> - [ASTERISK-25665
> <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] -
>
> Duplicate logging in queue log for EXITEMPTY events
> (Reported by Ove Aursand)
>
> - [ASTERISK-29043
> <https://issues.asterisk.org/jira/browse/ASTERISK-29043>] -
>
> app_queue: Leave empty sometimes not recorded as abandoned
> (Reported by Kfir Itzhak)
>
> - [ASTERISK-29042
> <https://issues.asterisk.org/jira/browse/ASTERISK-29042>] -
>
> res_parking: Parker UUID is no longer copied
> (Reported by Misha Vodsedalek)
>
> - [ASTERISK-28878
> <https://issues.asterisk.org/jira/browse/ASTERISK-28878>] -
>
> chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
> (Reported by Joseph Ades)
>
> - [ASTERISK-29046
> <https://issues.asterisk.org/jira/browse/ASTERISK-29046>] -
>
> 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)
>
> - [ASTERISK-29040
> <https://issues.asterisk.org/jira/browse/ASTERISK-29040>] -
>
> res_speech: Assertion on format
> (Reported by Nickolay V. Shmyrev)
>
> - [ASTERISK-29001
> <https://issues.asterisk.org/jira/browse/ASTERISK-29001>] -
>
> chan_pjsip does not process or forward 181 responses
> (Reported by Torrey Searle)
>
> - [ASTERISK-29034
> <https://issues.asterisk.org/jira/browse/ASTERISK-29034>] -
>
> Lastpause of realtime members is reseting
> (Reported by Evandro César Arruda)
>
> - [ASTERISK-27273
> <https://issues.asterisk.org/jira/browse/ASTERISK-27273>] -
>
> app_voicemail: When a voicemail is marked as "Urgent", it is not sent by
> email/processed by the mailcmd command
> (Reported by Leandro Dardini)
>
> - [ASTERISK-29033
> <https://issues.asterisk.org/jira/browse/ASTERISK-29033>] -
>
> res_pjsip_session: Aggressively terminates session on failed re-INVITE
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28974
> <https://issues.asterisk.org/jira/browse/ASTERISK-28974>] -
>
> res_rtp_asterisk: T.140 messages have appended RTP string to each message
> block.
> (Reported by Thomas Johnson)
>
> - [ASTERISK-29011
> <https://issues.asterisk.org/jira/browse/ASTERISK-29011>] -
>
> chan_sip: ToHost property not cleared on reload
> (Reported by Dennis)
>
> - [ASTERISK-29021
> <https://issues.asterisk.org/jira/browse/ASTERISK-29021>] -
>
> [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
> (Reported by cmaj)
>
> - [ASTERISK-28927
> <https://issues.asterisk.org/jira/browse/ASTERISK-28927>] -
>
> Asterisk crash in music on hold
> (Reported by David Cunningham)
>
> - [ASTERISK-28973
> <https://issues.asterisk.org/jira/browse/ASTERISK-28973>] -
>
> 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)
>
> - [ASTERISK-28995
> <https://issues.asterisk.org/jira/browse/ASTERISK-28995>] -
>
> res_pjsip_registrar: Expires on statically configured contacts is not
> correct
> (Reported by tootai)
>
> - [ASTERISK-28987
> <https://issues.asterisk.org/jira/browse/ASTERISK-28987>] -
>
> BridgeCreated ARI event shows wrong video_mode info
> (Reported by sungtae kim)
>
> - [ASTERISK-28978
> <https://issues.asterisk.org/jira/browse/ASTERISK-28978>] -
>
> acl: named_acl rule misconfiguration results in segfault on reading rule
> from realtime
> (Reported by Andrew Yager)
>
> - [ASTERISK-28975
> <https://issues.asterisk.org/jira/browse/ASTERISK-28975>] -
>
> res_http_websocket: Text payload data doesn't necessary include trailing
> zero
> (Reported by Nickolay V. Shmyrev)
>
> - [ASTERISK-28951
> <https://issues.asterisk.org/jira/browse/ASTERISK-28951>] -
>
> Inconsistent behaviour queues.conf when there is (not) a [general] section
> (Reported by Walter Doekes)
>
> - [ASTERISK-28965
> <https://issues.asterisk.org/jira/browse/ASTERISK-28965>] -
>
> res_pjsip: Apply outbound proxy to static contacts on AOR
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28930
> <https://issues.asterisk.org/jira/browse/ASTERISK-28930>] -
>
> ./configure --without-ssl build failure
> (Reported by Jaco Kroon)
>
> - [ASTERISK-28957
> <https://issues.asterisk.org/jira/browse/ASTERISK-28957>] -
>
> chan_sip: chan_sip does not process 400 response to an INVITE.
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28886
> <https://issues.asterisk.org/jira/browse/ASTERISK-28886>] -
>
> chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
> (Reported by Jared Smith)
>
> - [ASTERISK-28888
> <https://issues.asterisk.org/jira/browse/ASTERISK-28888>] -
>
> res_corosync: causes asterisk crash in huge distributed environment.
> (Reported by Università di Bologna - CESIA VoIP)
>
> - [ASTERISK-28954
> <https://issues.asterisk.org/jira/browse/ASTERISK-28954>] -
>
> StreamEcho() only returns 1 active stream
> (Reported by Bill Kervaski)
>
> - [ASTERISK-28955
> <https://issues.asterisk.org/jira/browse/ASTERISK-28955>] -
>
> "setvar" doesn't work properly in dahdi-channels.conf
> (Reported by Marin Odrljin)
>
> - [ASTERISK-28953
> <https://issues.asterisk.org/jira/browse/ASTERISK-28953>] -
>
> res_pjsip_session: Preserve stream label
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28942
> <https://issues.asterisk.org/jira/browse/ASTERISK-28942>] -
>
> res_sorcery_memory_cache: Individual object expiration behaves
> unexpectedly with full backend caching
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28950
> <https://issues.asterisk.org/jira/browse/ASTERISK-28950>] -
>
> Stale code in app_queue to check untouched channel
> (Reported by Walter Doekes)
>
> - [ASTERISK-28644
> <https://issues.asterisk.org/jira/browse/ASTERISK-28644>] -
>
> Stale comment in app_queue about ring_entry exception
> (Reported by Walter Doekes)
>
> - [ASTERISK-28952
> <https://issues.asterisk.org/jira/browse/ASTERISK-28952>] -
>
> Queue wrapuptime sometimes not respected (based on stale lastcall time)
> (Reported by Walter Doekes)
>
> - [ASTERISK-28938
> <https://issues.asterisk.org/jira/browse/ASTERISK-28938>] -
>
> core_unreal / core_local: Add support for multistream and re-negotiation
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28948
> <https://issues.asterisk.org/jira/browse/ASTERISK-28948>] -
>
> ARI channel create doesn't referencing the channel_id parameter
> (Reported by sungtae kim)
>
> - [ASTERISK-28939
> <https://issues.asterisk.org/jira/browse/ASTERISK-28939>] -
>
> res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28944
> <https://issues.asterisk.org/jira/browse/ASTERISK-28944>] -
>
> bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly
> doesn't re-negotiation
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28923
> <https://issues.asterisk.org/jira/browse/ASTERISK-28923>] -
>
> T.38 Segfaults in chan_pjsip_queryoption
> (Reported by Yury Kirsanov)
>
> - [ASTERISK-28940
> <https://issues.asterisk.org/jira/browse/ASTERISK-28940>] -
>
> /channels/create doesn't get any parameters from the body
> (Reported by sungtae kim)
>
> - [ASTERISK-28936
> <https://issues.asterisk.org/jira/browse/ASTERISK-28936>] -
>
> res_pjsip: crash when dialing non-sip uri
> (Reported by Walter Doekes)
>
> - [ASTERISK-28900
> <https://issues.asterisk.org/jira/browse/ASTERISK-28900>] -
>
> res_fax: Double frame free when gateway in use with off-nominal format
> usage
> (Reported by Gregory Massel)
>
> - [ASTERISK-28929
> <https://issues.asterisk.org/jira/browse/ASTERISK-28929>] -
>
> pjproject_bundled: Honor --without-pjproject.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28932
> <https://issues.asterisk.org/jira/browse/ASTERISK-28932>] -
>
> res_pjsip_logger writing too big packets
> (Reported by nappsoft)
>
> - [ASTERISK-28920
> <https://issues.asterisk.org/jira/browse/ASTERISK-28920>] -
>
> bridge show all causes crash
> (Reported by sungtae kim)
>
> - [ASTERISK-28921
> <https://issues.asterisk.org/jira/browse/ASTERISK-28921>] -
>
> Wrong return value check for fwrite when writing to pcap file
> (Reported by nappsoft)
>
> - [ASTERISK-28794
> <https://issues.asterisk.org/jira/browse/ASTERISK-28794>] -
>
> res_pjsip: Crash when escaping during URI printing
> (Reported by nappsoft)
>
> - [ASTERISK-28884
> <https://issues.asterisk.org/jira/browse/ASTERISK-28884>] -
>
> x-ast-orig-host not filtered out from request URI and To header
> (Reported by nappsoft)
>
> - [ASTERISK-28871
> <https://issues.asterisk.org/jira/browse/ASTERISK-28871>] -
>
> res_pjsip_session: Unnecessary re-Invite on call answer
> (Reported by Alexei Gradinari)
>
> - [ASTERISK-28903
> <https://issues.asterisk.org/jira/browse/ASTERISK-28903>] -
>
> res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28898
> <https://issues.asterisk.org/jira/browse/ASTERISK-28898>] -
>
> bridge_softmix: Conference bridge not passing silent rtp packets
> (Reported by Jonathan Hunter)
>
> - [ASTERISK-28892
> <https://issues.asterisk.org/jira/browse/ASTERISK-28892>] -
>
> res_musiconhold: Module res_musiconhold throws false warning
> (Reported by Nicholas John Koch)
>
> - [ASTERISK-28904
> <https://issues.asterisk.org/jira/browse/ASTERISK-28904>] -
>
> RTP ICE leaks the memory
> (Reported by sungtae kim)
>
> - [ASTERISK-26780
> <https://issues.asterisk.org/jira/browse/ASTERISK-26780>] -
>
> res_pjsip: PJSIP Registration Fails when transport=transport-udp6
> (Reported by Peter Sokolov)
>
> - [ASTERISK-28854
> <https://issues.asterisk.org/jira/browse/ASTERISK-28854>] -
>
> SIGSEGV when pjsip show history encounters IPV6 address
> (Reported by Roger James)
>
> - [ASTERISK-28797
> <https://issues.asterisk.org/jira/browse/ASTERISK-28797>] -
>
> [patch] tcptls: Fix notice when TLS is enabled but not configured.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28804
> <https://issues.asterisk.org/jira/browse/ASTERISK-28804>] -
>
> [patch] app_osplookup.c: Avoid a format truncation.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28776
> <https://issues.asterisk.org/jira/browse/ASTERISK-28776>] -
>
> Non async-signal-safe syscalls used after fork before exec
> (Reported by nappsoft)
>
> - [ASTERISK-28870
> <https://issues.asterisk.org/jira/browse/ASTERISK-28870>] -
>
> streams: One memory leak and one issue cloning streams
> (Reported by George Joseph)
>
> - [ASTERISK-28829
> <https://issues.asterisk.org/jira/browse/ASTERISK-28829>] -
>
> app_queue: leaking stasis subscription when Redirecting call
> (Reported by lvl)
>
> - [ASTERISK-25844
> <https://issues.asterisk.org/jira/browse/ASTERISK-25844>] -
>
> app_queue: Ghost channels in "core show channels" output
> (Reported by Etienne Lessard)
>
> - [ASTERISK-28859
> <https://issues.asterisk.org/jira/browse/ASTERISK-28859>] -
>
> pjsip: Increase maximum candidate count
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-22920
> <https://issues.asterisk.org/jira/browse/ASTERISK-22920>] -
>
> Crash while Forwarding from TLS extension with CHANNEL args
> secure_bridge_media and secure_bridge_signaling
> (Reported by Shlomi Gutman)
>
> - [ASTERISK-28852
> <https://issues.asterisk.org/jira/browse/ASTERISK-28852>] -
>
> Unprotected access to nochecksums variable, causes build failures
> (Reported by Guido Falsi)
>
> - [ASTERISK-28848
> <https://issues.asterisk.org/jira/browse/ASTERISK-28848>] -
>
> app_fax: Compile.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28846
> <https://issues.asterisk.org/jira/browse/ASTERISK-28846>] -
>
> stream: Enforce formats immutability
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28847
> <https://issues.asterisk.org/jira/browse/ASTERISK-28847>] -
>
> ARI channels cuts the endpoint string over 80 characters
> (Reported by sungtae kim)
>
> - [ASTERISK-28811
> <https://issues.asterisk.org/jira/browse/ASTERISK-28811>] -
>
> Crash occurs when fax session switches from T.38 to audio
> (Reported by Alexey Vasilyev)
>
> - [ASTERISK-28839
> <https://issues.asterisk.org/jira/browse/ASTERISK-28839>] -
>
> Sporadic crashes with Segmentation fault
> (Reported by Joeran Vinzens)
>
> - [ASTERISK-28835
> <https://issues.asterisk.org/jira/browse/ASTERISK-28835>] -
>
> IPv6 addresses in SDP incorrectly formatted
> (Reported by Daniel Heckl)
>
> - [ASTERISK-28372
> <https://issues.asterisk.org/jira/browse/ASTERISK-28372>] -
>
> Asterisk REPLY Wrong Contact header port (TCP)
> (Reported by Anton Satskiy)
>
> - [ASTERISK-24428
> <https://issues.asterisk.org/jira/browse/ASTERISK-24428>] -
>
> 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)
>
> - [ASTERISK-28838
> <https://issues.asterisk.org/jira/browse/ASTERISK-28838>] -
>
> AST_MODULE_INFO requires, MODULEINFO does not mention
> (Reported by Alexander Traud)
>
> - [ASTERISK-28841
> <https://issues.asterisk.org/jira/browse/ASTERISK-28841>] -
>
> app_confbridge: Add support for disabling text messaging for a user
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28837
> <https://issues.asterisk.org/jira/browse/ASTERISK-28837>] -
>
> pjproject_bundled: Honor --without-pjproject.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28827
> <https://issues.asterisk.org/jira/browse/ASTERISK-28827>] -
>
> res_rtp_asterisk: Loop when receive buffer is flushed by a received packet
> that is also in receive buffer with NACK
> (Reported by nappsoft)
>
> - [ASTERISK-27195
> <https://issues.asterisk.org/jira/browse/ASTERISK-27195>] -
>
> chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
> (Reported by Joshua Roys)
>
> - [ASTERISK-28826
> <https://issues.asterisk.org/jira/browse/ASTERISK-28826>] -
>
> res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
> (Reported by nappsoft)
>
> - [ASTERISK-28812
> <https://issues.asterisk.org/jira/browse/ASTERISK-28812>] -
>
> First DTMF is not get
> (Reported by Bernard Merindol)
>
> - [ASTERISK-28758
> <https://issues.asterisk.org/jira/browse/ASTERISK-28758>] -
>
> pjsip startup errors when using "with-ssl" configure option
> (Reported by Patrick Wakano)
>
> - [ASTERISK-28824
> <https://issues.asterisk.org/jira/browse/ASTERISK-28824>] -
>
> BuildSystem: Search for Python/C API when possibly needed only.
> (Reported by Alexander Traud)
>
> - [ASTERISK-27717
> <https://issues.asterisk.org/jira/browse/ASTERISK-27717>] -
>
> [patch] BuildSystem: In NetBSD, the Python Programming Language is
> python-2.7.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28817
> <https://issues.asterisk.org/jira/browse/ASTERISK-28817>] -
>
> chan_pjsip: constant DTMF tone if RTP is not setup yet
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28819
> <https://issues.asterisk.org/jira/browse/ASTERISK-28819>] -
>
> [patch] bridge_softmix_binaural: Show state in menuselect.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28816
> <https://issues.asterisk.org/jira/browse/ASTERISK-28816>] -
>
> [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28818
> <https://issues.asterisk.org/jira/browse/ASTERISK-28818>] -
>
> [patch] BuildSystem: Allow space in path.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28809
> <https://issues.asterisk.org/jira/browse/ASTERISK-28809>] -
>
> [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28796
> <https://issues.asterisk.org/jira/browse/ASTERISK-28796>] -
>
> func_channel: cannot read fields exten, context, userfield, channame from
> dialplan
> (Reported by Sébastien Duthil)
>
> - [ASTERISK-28803
> <https://issues.asterisk.org/jira/browse/ASTERISK-28803>] -
>
> [patch] chan_unistim: Avoid tautological warnings with clang.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28808
> <https://issues.asterisk.org/jira/browse/ASTERISK-28808>] -
>
> [patch] test_stasis: Avoid always true warning with clang.
> (Reported by Alexander Traud)
>
> - [ASTERISK-28056
> <https://issues.asterisk.org/jira/browse/ASTERISK-28056>] -
>
> res_pjsip: Incorrect endpoint status after endpoint synchronization for a
> specific AOR
> (Reported by Jason Hord)
>
> - [ASTERISK-28795
> <https://issues.asterisk.org/jira/browse/ASTERISK-28795>] -
>
> channel: write to a stream on multi-frame writes
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28789
> <https://issues.asterisk.org/jira/browse/ASTERISK-28789>] -
>
> test_utils: incorrectly printing error 'declined to load'
> (Reported by Alexander Traud)
>
> - [ASTERISK-28788
> <https://issues.asterisk.org/jira/browse/ASTERISK-28788>] -
>
> func_aes: incorrectly printing error 'declined to load'
> (Reported by Alexander Traud)
>
> - [ASTERISK-28790
> <https://issues.asterisk.org/jira/browse/ASTERISK-28790>] -
>
> Crash during conference call using confbridge and video
> (Reported by Pascal Cadotte Michaud)
>
> - [ASTERISK-16676
> <https://issues.asterisk.org/jira/browse/ASTERISK-16676>] -
>
> DAHDIRAS fails to properly initiate pppd unless asterisk is running as root
> (Reported by Jaco Kroon)
>
> - [ASTERISK-21205
> <https://issues.asterisk.org/jira/browse/ASTERISK-21205>] -
>
> [patch] dundi_read_result crash due to negative number
> (Reported by Jaco Kroon)
>
> - [ASTERISK-28784
> <https://issues.asterisk.org/jira/browse/ASTERISK-28784>] -
>
> res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28743
> <https://issues.asterisk.org/jira/browse/ASTERISK-28743>] -
>
> Asterisk is crashing if the 200 OK with SDP
> (Reported by sungtae kim)
>
> - [ASTERISK-28783
> <https://issues.asterisk.org/jira/browse/ASTERISK-28783>] -
>
> res_pjsip_session: Allow default non-audio streams to have reflected state
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28774
> <https://issues.asterisk.org/jira/browse/ASTERISK-28774>] -
>
> chan_pjsip's rtptimeout is erroneously triggered during direct-media
> (native_rtp) bridge
> (Reported by Michael Neuhauser)
>
> - [ASTERISK-20325
> <https://issues.asterisk.org/jira/browse/ASTERISK-20325>] -
>
> Comments in configs/func_odbc.conf.sample are not consistent with
> examples. Missing examples.
> (Reported by Olivier Krief)
>
> - [ASTERISK-28780
> <https://issues.asterisk.org/jira/browse/ASTERISK-28780>] -
>
> app_mixmonitor: Memory leak due to race condition between AMI MixMonitor
> and hangup
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28773
> <https://issues.asterisk.org/jira/browse/ASTERISK-28773>] -
>
> Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
> (Reported by Torrey Searle)
>
> - [ASTERISK-28769
> <https://issues.asterisk.org/jira/browse/ASTERISK-28769>] -
>
> DTLS Handshake Fails to Occur if ice_support is enabled but not used
> (Reported by Torrey Searle)
>
> - [ASTERISK-28759
> <https://issues.asterisk.org/jira/browse/ASTERISK-28759>] -
>
> A non negotiated rtp frame causes call disconnection when there is a SSRC
> change
> (Reported by Paulo Vicentini)
>
> - [ASTERISK-26711
> <https://issues.asterisk.org/jira/browse/ASTERISK-26711>] -
>
> func_enum: ENUM code wrong case
> (Reported by Vitold)
>
> - [ASTERISK-23407
> <https://issues.asterisk.org/jira/browse/ASTERISK-23407>] -
>
> Fix the FSF address in the headers of lots of pjproject files
> (Reported by Jared Smith)
>
> - [ASTERISK-19460
> <https://issues.asterisk.org/jira/browse/ASTERISK-19460>] -
>
> [patch] Function TXTCIDNAME never actually makes DNS calls and always
> returns an empty string
> (Reported by George Joseph)
>
> - [ASTERISK-28766
> <https://issues.asterisk.org/jira/browse/ASTERISK-28766>] -
>
> PJSIP blind transfer not completed after using Proceeding()
> (Reported by lvl)
>
> - [ASTERISK-28764
> <https://issues.asterisk.org/jira/browse/ASTERISK-28764>] -
>
> res_rtp_asterisk: Improve NACK support and seqno handling
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28755
> <https://issues.asterisk.org/jira/browse/ASTERISK-28755>] -
>
> SIP/Stasis: SIP headers not transmitted in the "variables" field
> (Reported by Jean Aunis - Prescom)
>
> - [ASTERISK-28685
> <https://issues.asterisk.org/jira/browse/ASTERISK-28685>] -
>
> check_expr2: linking (when hardening) and cross-compiling troubles
> (Reported by Sebastian Kemper)
>
> - [ASTERISK-28754
> <https://issues.asterisk.org/jira/browse/ASTERISK-28754>] -
>
> ASTERISK-28738 Causes Audio Issue After Hold
> (Reported by Ross Beer)
>
> - [ASTERISK-28697
> <https://issues.asterisk.org/jira/browse/ASTERISK-28697>] -
>
> res_pjsip: Named ACL does not update on reload if changed
> (Reported by Timothy Vanderaerden)
>
> - [ASTERISK-28746
> <https://issues.asterisk.org/jira/browse/ASTERISK-28746>] -
>
> res_pjsip_outbound_registration keeps retrying the first entry in a SRV
> record set
> (Reported by George Joseph)
>
> - [ASTERISK-28716
> <https://issues.asterisk.org/jira/browse/ASTERISK-28716>] -
>
> ICE: pjnath shouldn't wait for ICE to complete before allowing sending
> (Reported by Benjamin Keith Ford)
>
> - [ASTERISK-28738
> <https://issues.asterisk.org/jira/browse/ASTERISK-28738>] -
>
> Incorrect state machine used when MOH_PASSTHRU is used
> (Reported by Torrey Searle)
>
> - [ASTERISK-28742
> <https://issues.asterisk.org/jira/browse/ASTERISK-28742>] -
>
> res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28735
> <https://issues.asterisk.org/jira/browse/ASTERISK-28735>] -
>
> Realtime MoH Unknown format '' -- defaulting to SLIN
> (Reported by Ross Beer)
>
> - [ASTERISK-28730
> <https://issues.asterisk.org/jira/browse/ASTERISK-28730>] -
>
> res_pjsip_session: Fix out of order session refreshes
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-26955
> <https://issues.asterisk.org/jira/browse/ASTERISK-26955>] -
>
> pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited
> by "[]" Rejected
> (Reported by Peter Sokolov)
>
> - [ASTERISK-28718
> <https://issues.asterisk.org/jira/browse/ASTERISK-28718>] -
>
> chan_sip: Returns 403 if RTP ports are depleted, should return 503
> (Reported by Walter Doekes)
>
> - [ASTERISK-28713
> <https://issues.asterisk.org/jira/browse/ASTERISK-28713>] -
>
> res_stasis_playback: Error building JSON
> (Reported by Sébastien Duthil)
>
> - [ASTERISK-28714
> <https://issues.asterisk.org/jira/browse/ASTERISK-28714>] -
>
> REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759)
> Causes Segfaults
> (Reported by Ross Beer)
>
> - [ASTERISK-26082
> <https://issues.asterisk.org/jira/browse/ASTERISK-26082>] -
>
> res_pjsip_messaging: MessageSend Content-Type can't be changed
> (Reported by Alex)
>
> - [ASTERISK-28423
> <https://issues.asterisk.org/jira/browse/ASTERISK-28423>] -
>
> ARI causes STASIS Deadlock
> (Reported by Ross Beer)
>
> - [ASTERISK-28679
> <https://issues.asterisk.org/jira/browse/ASTERISK-28679>] -
>
> stasis application is destroyed after its creation
> (Reported by Francois Blackburn)
>
> - [ASTERISK-25421
> <https://issues.asterisk.org/jira/browse/ASTERISK-25421>] -
>
> PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when
> sending
> (Reported by Dmitriy Serov)
>
> - [ASTERISK-28686
> <https://issues.asterisk.org/jira/browse/ASTERISK-28686>] -
>
> chan_sip strictrtp=yes fails when media source is changed: no audio
> (Reported by Walter Doekes)
>
> - [ASTERISK-28139
> <https://issues.asterisk.org/jira/browse/ASTERISK-28139>] -
>
> RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
> (Reported by Paul Brooks)
>
> - [ASTERISK-28677
> <https://issues.asterisk.org/jira/browse/ASTERISK-28677>] -
>
> CDR billsec is always 0 for transferred calls
> (Reported by Maciej Michno)
>
> - [ASTERISK-28702
> <https://issues.asterisk.org/jira/browse/ASTERISK-28702>] -
>
> chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
> (Reported by Andrew Siplas)
>
> - [ASTERISK-24484
> <https://issues.asterisk.org/jira/browse/ASTERISK-24484>] -
>
> Update documentation for statsd module - usage requirements unclear
> (Reported by Dan Jenkins)
>
> - [ASTERISK-28706
> <https://issues.asterisk.org/jira/browse/ASTERISK-28706>] -
>
> silk 24hHz doesn't show up in 'core show translation' output
> (Reported by Sean Bright)
>
> - [ASTERISK-28695
> <https://issues.asterisk.org/jira/browse/ASTERISK-28695>] -
>
> core: minmemfree watermark uses free RAM, not available RAM
> (Reported by Kevin Flyn)
>
> - [ASTERISK-28693
> <https://issues.asterisk.org/jira/browse/ASTERISK-28693>] -
>
> chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the
> dialplan
> (Reported by Frank Matano)
>
> - [ASTERISK-23739
> <https://issues.asterisk.org/jira/browse/ASTERISK-23739>] -
>
> [patch]Segfault forwarding voicemail with ODBC storage enabled and
> realtime voicemail_data is used
> (Reported by Stas Kobzar)
>
> - [ASTERISK-27622
> <https://issues.asterisk.org/jira/browse/ASTERISK-27622>] -
>
> empty voicemail.conf required for ARA (realtime) voicemail to leave message
> (Reported by Jim Van Meggelen)
>
> - [ASTERISK-21794
> <https://issues.asterisk.org/jira/browse/ASTERISK-21794>] -
>
> CLI command 'realtime update2' syntax failure when using according to
> usage help
> (Reported by Cedric BASSAGET)
>
> - [ASTERISK-28349
> <https://issues.asterisk.org/jira/browse/ASTERISK-28349>] -
>
> Pause reason not reported in QueueMember AMI event
> (Reported by Niksa Baldun)
>
> - [ASTERISK-25429
> <https://issues.asterisk.org/jira/browse/ASTERISK-25429>] -
>
> res_pjsip_endpoint_identifier_ip: Document support for hostnames
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-27775
> <https://issues.asterisk.org/jira/browse/ASTERISK-27775>] -
>
> res_pjsip_notify: Multiple Event headers can be present instead of just one
> (Reported by AvayaXAsterisk)
>
> - [ASTERISK-28682
> <https://issues.asterisk.org/jira/browse/ASTERISK-28682>] -
>
> app_record: Lack of `beep` audio file causes application to return error
> and hangup
> (Reported by Corey Farrell)
>
> - [ASTERISK-28507
> <https://issues.asterisk.org/jira/browse/ASTERISK-28507>] -
>
> Wiki docs missing for MessageWaiting
> (Reported by David M. Lee)
>
> - [ASTERISK-27759
> <https://issues.asterisk.org/jira/browse/ASTERISK-27759>] -
>
> res_pjsip_pubsub: Subscription persistence does not preserve XML version
> number
> (Reported by Bryan Nelson)
>
> - [ASTERISK-28605
> <https://issues.asterisk.org/jira/browse/ASTERISK-28605>] -
>
> chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show
> span X
> (Reported by Dirk Wendland)
>
> - [ASTERISK-28633
> <https://issues.asterisk.org/jira/browse/ASTERISK-28633>] -
>
> stasis bridge topic leak
> (Reported by Joeran Vinzens)
>
> - [ASTERISK-28492
> <https://issues.asterisk.org/jira/browse/ASTERISK-28492>] -
>
> pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
> (Reported by Jean-Denis Girard)
>
> - [ASTERISK-28562
> <https://issues.asterisk.org/jira/browse/ASTERISK-28562>] -
>
> SIP WSS message not processed until next frame arrives
> (Reported by Robert Sutton)
>
> - [ASTERISK-28667
> <https://issues.asterisk.org/jira/browse/ASTERISK-28667>] -
>
> Asterisk ignores parsing of config files if a Byte order mark is present
> (Reported by Robin Leffmann)
>
> - [ASTERISK-28625
> <https://issues.asterisk.org/jira/browse/ASTERISK-28625>] -
>
> Playback of local files impacted by large media cache
> (Reported by Kevin Reeves)
>
> - [ASTERISK-27243
> <https://issues.asterisk.org/jira/browse/ASTERISK-27243>] -
>
> contrib: valgrind.supp doesn't suppress what it's supposed to due to
> invalid syntax
> (Reported by Richard Kenner)
>
> - [ASTERISK-28664
> <https://issues.asterisk.org/jira/browse/ASTERISK-28664>] -
>
> "trustrpid" is misspelled in sip_to_pjsip.py
> (Reported by Pascal Cadotte Michaud)
>
> - [ASTERISK-28636
> <https://issues.asterisk.org/jira/browse/ASTERISK-28636>] -
>
> app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28604
> <https://issues.asterisk.org/jira/browse/ASTERISK-28604>] -
>
> app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
> (Reported by George Joseph)
>
> - [ASTERISK-28659
> <https://issues.asterisk.org/jira/browse/ASTERISK-28659>] -
>
> res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs
> create additional streams and offer does not have them
> (Reported by nappsoft)
>
> - [ASTERISK-28660
> <https://issues.asterisk.org/jira/browse/ASTERISK-28660>] -
>
> res_fax: wrap Asterisk initiated negotiation with config option
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28626
> <https://issues.asterisk.org/jira/browse/ASTERISK-28626>] -
>
> Missing arguments in PJSIP_CONTACT function documentation
> (Reported by Pascal Cadotte Michaud)
>
> - [ASTERISK-28609
> <https://issues.asterisk.org/jira/browse/ASTERISK-28609>] -
>
> Memory Leak in res_rtp_asterisk.c
> (Reported by Ted G)
>
> - [ASTERISK-28651
> <https://issues.asterisk.org/jira/browse/ASTERISK-28651>] -
>
> chan_sip logs errors on tx to non-existent TCP connections
> (Reported by Jaco Kroon)
>
> - [ASTERISK-28502
> <https://issues.asterisk.org/jira/browse/ASTERISK-28502>] -
>
> chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
> (Reported by Ross Beer)
>
> - [ASTERISK-28641
> <https://issues.asterisk.org/jira/browse/ASTERISK-28641>] -
>
> res_pjsip Segfaults when realtime configuration to an AOR points to a not
> existent AOR
> (Reported by Ross Beer)
>
> - [ASTERISK-28647
> <https://issues.asterisk.org/jira/browse/ASTERISK-28647>] -
>
> chan_sip: RTP frames not transmitted after emitting a COLP
> (Reported by Jean Aunis - Prescom)
>
> - [ASTERISK-28637
> <https://issues.asterisk.org/jira/browse/ASTERISK-28637>] -
>
> chan_sip+native_bridge_rtp: directmedia compatibility check failure when
> negociated ptime is not default ptime.
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28445
> <https://issues.asterisk.org/jira/browse/ASTERISK-28445>] -
>
> res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when
> TEST_FRAMEWORK enabled
> (Reported by Bernhard Schmidt)
>
> - [ASTERISK-28631
> <https://issues.asterisk.org/jira/browse/ASTERISK-28631>] -
>
> res_parking: Doesn't park when parkee and parker are the same
> (Reported by Ross Beer)
>
> - [ASTERISK-28621
> <https://issues.asterisk.org/jira/browse/ASTERISK-28621>] -
>
> Enforce T.38 error correction mode at 200 ok received
> (Reported by Salah Ahmed)
>
> - [ASTERISK-28624
> <https://issues.asterisk.org/jira/browse/ASTERISK-28624>] -
>
> res_pjsip_outbound_registration: add SRV failover
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28608
> <https://issues.asterisk.org/jira/browse/ASTERISK-28608>] -
>
> app_amd: Use time calculation to calculate timeout
> (Reported by Michael Cargile)
>
> - [ASTERISK-28615
> <https://issues.asterisk.org/jira/browse/ASTERISK-28615>] -
>
> chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28576
> <https://issues.asterisk.org/jira/browse/ASTERISK-28576>] -
>
> res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't
> match
> (Reported by Joshua Elson)
>
> - [ASTERISK-26481
> <https://issues.asterisk.org/jira/browse/ASTERISK-26481>] -
>
> FILE function grabs garbage along with read data when target line has no
> newline
> (Reported by Jonathan Harris)
>
> - [ASTERISK-28618
> <https://issues.asterisk.org/jira/browse/ASTERISK-28618>] -
>
> bridge_softmix: hold not cleared when joining a softmix bridge
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28616
> <https://issues.asterisk.org/jira/browse/ASTERISK-28616>] -
>
> parking: Deadlock when multi call parking
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28572
> <https://issues.asterisk.org/jira/browse/ASTERISK-28572>] -
>
> Memory leaks in res_calendar_exchange and res_calendar_icalendar
> (Reported by Yoooooo Ha)
>
> - [ASTERISK-28585
> <https://issues.asterisk.org/jira/browse/ASTERISK-28585>] -
>
> ari/resource_events: Crash in event session cleanup
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28590
> <https://issues.asterisk.org/jira/browse/ASTERISK-28590>] -
>
> utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid
> argument"
> (Reported by Speed Dial Dave)
>
> - [ASTERISK-28578
> <https://issues.asterisk.org/jira/browse/ASTERISK-28578>] -
>
> race condition on pjsip channelstats command
> (Reported by Salah Ahmed)
>
> - [ASTERISK-28571
> <https://issues.asterisk.org/jira/browse/ASTERISK-28571>] -
>
> cdr_pgsql: accesses obsolete (and finally removed) column
> (Reported by Christoph Moench-Tegeder)
>
> - [ASTERISK-28575
> <https://issues.asterisk.org/jira/browse/ASTERISK-28575>] -
>
> MWI Send Notify Crash on 16.6
> (Reported by Joshua Elson)
>
> - [ASTERISK-28574
> <https://issues.asterisk.org/jira/browse/ASTERISK-28574>] -
>
> pjproject fails to build on 16.6.0, works on 16.5
> (Reported by Niklas Larsson)
>
> - [ASTERISK-28561
> <https://issues.asterisk.org/jira/browse/ASTERISK-28561>] -
>
> Asterisk Deadlocks
> (Reported by Aheliotech)
>
> - [ASTERISK-28086
> <https://issues.asterisk.org/jira/browse/ASTERISK-28086>] -
>
> chan_pjsip: Crash when initiating PlayDTMF over AMI
> (Reported by Jeremiah Gadd)
>
> - [ASTERISK-28552
> <https://issues.asterisk.org/jira/browse/ASTERISK-28552>] -
>
> res_pjsip_mwi: Frack during unload on unsolicited_mwi container
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28566
> <https://issues.asterisk.org/jira/browse/ASTERISK-28566>] -
>
> CDR backend unload problem during active call(s)
> (Reported by Marian Piater)
>
> - [ASTERISK-28553
> <https://issues.asterisk.org/jira/browse/ASTERISK-28553>] -
>
> stasis.c: Crash during unload
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28544
> <https://issues.asterisk.org/jira/browse/ASTERISK-28544>] -
>
> Wrong contact representation in ipv6 mode
> (Reported by Jørgen H)
>
> - [ASTERISK-28534
> <https://issues.asterisk.org/jira/browse/ASTERISK-28534>] -
>
> Segmentation fault when there is no priority for an extension
> (Reported by Timothy Vanderaerden)
>
> - [ASTERISK-28463
> <https://issues.asterisk.org/jira/browse/ASTERISK-28463>] -
>
> res_pjsip_path: Crash when invalid contact is configured
> (Reported by Juan Martin)
>
> - [ASTERISK-28521
> <https://issues.asterisk.org/jira/browse/ASTERISK-28521>] -
>
> pjsip: Memory Leak
> (Reported by Mark)
>
> - [ASTERISK-28523
> <https://issues.asterisk.org/jira/browse/ASTERISK-28523>] -
>
> Asterisk 16.5.0 Memory leak
> (Reported by Cyril Ramière)
>
> - [ASTERISK-28536
> <https://issues.asterisk.org/jira/browse/ASTERISK-28536>] -
>
> Asterisk release candidates fail to build on FreeBSD
> (Reported by Guido Falsi)
>
> - [ASTERISK-28538
> <https://issues.asterisk.org/jira/browse/ASTERISK-28538>] -
>
> chan_pjsip: Deadlock on fax detection
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28497
> <https://issues.asterisk.org/jira/browse/ASTERISK-28497>] -
>
> func_odbc: truncating Unicode string on readsql
> (Reported by Boris P. Korzun)
>
> - [ASTERISK-23756
> <https://issues.asterisk.org/jira/browse/ASTERISK-23756>] -
>
> setvar directive when used in template and a child of said template,
> results in duplicate variable names
> (Reported by Michael Goryainov)
>
> - [ASTERISK-28527
> <https://issues.asterisk.org/jira/browse/ASTERISK-28527>] -
>
> ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28525
> <https://issues.asterisk.org/jira/browse/ASTERISK-28525>] -
>
> chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up
> (Reported by Frederic LE FOLL)
>
> - [ASTERISK-28511
> <https://issues.asterisk.org/jira/browse/ASTERISK-28511>] -
>
> codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32
> (Reported by Ruddy G)
>
> - [ASTERISK-28499
> <https://issues.asterisk.org/jira/browse/ASTERISK-28499>] -
>
> translate: Crash when frame does not have a "src" field set
> (Reported by Gregory Massel)
>
> - [ASTERISK-25592
> <https://issues.asterisk.org/jira/browse/ASTERISK-25592>] -
>
> chan_unistim: Clang Warning: variable sized type not at end of a struct
> (Reported by Alexander Traud)
>
> - [ASTERISK-28488
> <https://issues.asterisk.org/jira/browse/ASTERISK-28488>] -
>
> pjsip mwi: n+1 sip notify's sent on re-register
> (Reported by Chris Savinovich)
>
> - [ASTERISK-28509
> <https://issues.asterisk.org/jira/browse/ASTERISK-28509>] -
>
> PJSIP cnonce generated on Linux contains 36 characters, NEC only supports
> up to 32 characters
> (Reported by Dan Cropp)
>
> - [ASTERISK-28505
> <https://issues.asterisk.org/jira/browse/ASTERISK-28505>] -
>
> app_voicemail/IMAP: segfault in leave_voicemail because not checking
> mailstream
> (Reported by Alexei Gradinari)
>
> - [ASTERISK-28487
> <https://issues.asterisk.org/jira/browse/ASTERISK-28487>] -
>
> compile menuselect on gentoo
> (Reported by Kilburn)
>
> - [ASTERISK-28472
> <https://issues.asterisk.org/jira/browse/ASTERISK-28472>] -
>
> Asterisk occasionally passes a NULL as srtp->session to
> srtp_protect/unprotect causing SEGV
> (Reported by Jonas Swiatek)
>
> - [ASTERISK-28498
> <https://issues.asterisk.org/jira/browse/ASTERISK-28498>] -
>
> cel / cdr: Event times may be incorrect
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28480
> <https://issues.asterisk.org/jira/browse/ASTERISK-28480>] -
>
> json integer overflow in ssrc and timestamp
> (Reported by Salah Ahmed)
>
> - [ASTERISK-28228
> <https://issues.asterisk.org/jira/browse/ASTERISK-28228>] -
>
> res_pjsip: pjsip show contacts prints double entries
> (Reported by Ian Jones)
>
> - [ASTERISK-28483
> <https://issues.asterisk.org/jira/browse/ASTERISK-28483>] -
>
> packet lost on UDPTL wrap around
> (Reported by Torrey Searle)
>
> *Improvements made in this release:*
> -----------------------------------
>
> - [ASTERISK-28959
> <https://issues.asterisk.org/jira/browse/ASTERISK-28959>] -
>
> res_pjsip: Added option for disable rport parameter set
> (Reported by sungtae kim)
>
> - [ASTERISK-28958
> <https://issues.asterisk.org/jira/browse/ASTERISK-28958>] -
>
> Continue reading string when ping received by websocket
> (Reported by Nickolay V. Shmyrev)
>
> - [ASTERISK-28945
> <https://issues.asterisk.org/jira/browse/ASTERISK-28945>] -
>
> AMI SendText - add Content-Type parameter
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28949
> <https://issues.asterisk.org/jira/browse/ASTERISK-28949>] -
>
> res_http_websocket: Add masking to websocket client
> (Reported by Moises Silva)
>
> - [ASTERISK-28899
> <https://issues.asterisk.org/jira/browse/ASTERISK-28899>] -
>
> Upgrade Asterisk to bundled pjproject 2.10
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28895
> <https://issues.asterisk.org/jira/browse/ASTERISK-28895>] -
>
> res_pjsip_logger: Add tons'o'functionality
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28896
> <https://issues.asterisk.org/jira/browse/ASTERISK-28896>] -
>
> ari: Add support for specifying variables on channel create
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28879
> <https://issues.asterisk.org/jira/browse/ASTERISK-28879>] -
>
> pjproject has race conditions in it's build system
> (Reported by Guido Falsi)
>
> - [ASTERISK-28866
> <https://issues.asterisk.org/jira/browse/ASTERISK-28866>] -
>
> third-party/pjproject/configure.m4 contains bashisms
> (Reported by Guido Falsi)
>
> - [ASTERISK-28853
> <https://issues.asterisk.org/jira/browse/ASTERISK-28853>] -
>
> Missing include on FreeBSD
> (Reported by Guido Falsi)
>
> - [ASTERISK-28832
> <https://issues.asterisk.org/jira/browse/ASTERISK-28832>] -
>
> chan_mobile creates PCMA streams that make some VoIP clients crash or not
> render received audio
> (Reported by Peter Turczak)
>
> - [ASTERISK-28813
> <https://issues.asterisk.org/jira/browse/ASTERISK-28813>] -
>
> func_volume: Allow decimal numbers as parameter to improve granularity
> (Reported by Jean Aunis - Prescom)
>
> - [ASTERISK-28777
> <https://issues.asterisk.org/jira/browse/ASTERISK-28777>] -
>
> Codec Negotiation: add outgoing_call_offer_prefs option
> (Reported by Kevin Harwell)
>
> - [ASTERISK-27946
> <https://issues.asterisk.org/jira/browse/ASTERISK-27946>] -
>
> dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it
> shouldn't
> (Reported by Joshua Elson)
>
> - [ASTERISK-28782
> <https://issues.asterisk.org/jira/browse/ASTERISK-28782>] -
>
> Add support for Content-Disposition header in multi-part INVITES
> (Reported by Torrey Searle)
>
> - [ASTERISK-28787
> <https://issues.asterisk.org/jira/browse/ASTERISK-28787>] -
>
> res_pjsip_session: Decide more intelligently when to add video
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28756
> <https://issues.asterisk.org/jira/browse/ASTERISK-28756>] -
>
> Codec Negotiation: add incoming_call_offer_pref option
> (Reported by Kevin Harwell)
>
> - [ASTERISK-28750
> <https://issues.asterisk.org/jira/browse/ASTERISK-28750>] -
>
> TLS/SSL Key too small error
> (Reported by Martin Zeh)
>
> - [ASTERISK-28733
> <https://issues.asterisk.org/jira/browse/ASTERISK-28733>] -
>
> stream: Add support for adding/removing streams during SFU/calls
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-24798
> <https://issues.asterisk.org/jira/browse/ASTERISK-24798>] -
>
> Documentation - Clarify That Format Is Set By File Name Extension In
> MixMonitor
> (Reported by xrobau)
>
> - [ASTERISK-28726
> <https://issues.asterisk.org/jira/browse/ASTERISK-28726>] -
>
> install_prereq script uses the interactive mode when installing aptitude
> (Reported by Sylvain Afchain)
>
> - [ASTERISK-28710
> <https://issues.asterisk.org/jira/browse/ASTERISK-28710>] -
>
> Should be able to disable the /httpstatus URI in the built-in HTTP server
> (Reported by Sean Bright)
>
> - [ASTERISK-28484
> <https://issues.asterisk.org/jira/browse/ASTERISK-28484>] -
>
> Add AudioSocket support
> (Reported by Seán C. McCord)
>
> - [ASTERISK-28638
> <https://issues.asterisk.org/jira/browse/ASTERISK-28638>] -
>
> Simplify dialplan for Dial, Page, and ChanIsAvail
> (Reported by cmaj)
>
> - [ASTERISK-28673
> <https://issues.asterisk.org/jira/browse/ASTERISK-28673>] -
>
> GET FULL VARIABLE documentation clarification
> (Reported by Jonathan Harris)
>
> - [ASTERISK-28629
> <https://issues.asterisk.org/jira/browse/ASTERISK-28629>] -
>
> [patch] Add an "inhibitCOLP" flag to the bridges REST API
> (Reported by Jean Aunis - Prescom)
>
> - [ASTERISK-28658
> <https://issues.asterisk.org/jira/browse/ASTERISK-28658>] -
>
> app_confbridge: Add support for setting maximum sample rate
> (Reported by Joshua C. Colp)
>
> - [ASTERISK-28602
> <https://issues.asterisk.org/jira/browse/ASTERISK-28602>] -
>
> res_pjsip_outbound_registration: Maximum retries reached
> (Reported by Daniel)
>
> - [ASTERISK-28586
> <https://issues.asterisk.org/jira/browse/ASTERISK-28586>] -
>
> Typo in README-SERIOUSLY.bestpractices.md
> (Reported by Sam Banks)
>
> - [ASTERISK-22192
> <https://issues.asterisk.org/jira/browse/ASTERISK-22192>] -
>
> [patch] Allow voicemail forwards with ODBC backend when format differs
> from attachfmt column
> (Reported by cmaj)
>
> - [ASTERISK-28567
> <https://issues.asterisk.org/jira/browse/ASTERISK-28567>] -
>
> Problem with ASTERISK-20207: Asterisk should clear out any .lock files in
> the voice mail directory on startup.
> (Reported by Michael)
>
> - [ASTERISK-28542
> <https://issues.asterisk.org/jira/browse/ASTERISK-28542>] -
>
> [patch] add the ability for asterisk to generate on-hold re-invites
> (Reported by Torrey Searle)
>
> - [ASTERISK-28512
> <https://issues.asterisk.org/jira/browse/ASTERISK-28512>] -
>
> Add pass-through support for H.265 (HEVC) codec
> (Reported by Florian Floimair)
>
> For a full list of changes in this release, please see the ChangeLog:
> https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
>
> *Thank you for your continued support of Asterisk!*
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> To UNSUBSCRIBE or update options visit:
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--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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