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 < asteriskt...@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 -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev -- 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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