> 2020年10月28日 上午5:12，asterisk-dev-requ...@lists.digium.com 写道： > > Send asterisk-dev mailing list submissions to > asterisk-dev@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-dev > or, via email, send a message with subject or body 'help' to > asterisk-dev-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-dev-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-dev digest..." > > > Today's Topics: > > 1. Re: Problem with SDP session id in 200 OK during ReInvite > (Michael Maier) > 2. Re: Asterisk 18.0.0 Now Available (John Kiniston) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 27 Oct 2020 18:47:36 +0100 > From: Michael Maier <m1278...@mailbox.org> > To: asterisk-dev@lists.digium.com > Subject: Re: [asterisk-dev] Problem with SDP session id in 200 OK > during ReInvite > Message-ID: <6e2ceaf7-b482-0880-9e2c-648c408f0...@mailbox.org> > Content-Type: text/plain; charset=utf-8 > > Hello Joshua, > > > On 27.10.20 at 10:07 Joshua C. Colp wrote: >> On Mon, Oct 26, 2020 at 2:02 PM Michael Maier <m1278...@mailbox.org> wrote: >> >>> Hello! >>> >>> I'm facing the problem, that *sometimes* the SDP session ID isn't >>> incremented in the 200 OK, which asterisk sends as answer to a ReInvite it >>> got from the peer (use case: session >>> timer handling). This leads to broken calls, because the SDP session ID >>> must be incremented if the session description has changed (the session >>> description has changed). >>> >>> Modifying the SDP session ID is possible in >>> res/res_pjsip/pjsip_message_filter.c / filter_on_tx_message() for SDPs >>> contained in an Invite, which is created and sent by asterisk. >>> At the moment, I'm already modifying the SDP session ID at this place, >>> because of another problem: >>> >> >> <snip> >> >> >>> Maybe it's too late and the processing of the 200 OK doesn't hit this part >>> at all? >>> >>> Or is there any other possibility to modify the SDP session ID contained >>> in the 200 OK, that is sent by asterisk as an answer to a ReInvite? >>> >> >> It should be modifiable there, it injects itself at the transaction layer >> to be called for both requests and responses. I'm not sure under what >> scenarios (if any) it would not be called. > > thanks for your estimation! Meanwhile, I hope I have found the problem now. I > changed my workaround to always statically modify the SDP session id (the > condition seemed to be the > problem). But tests are going on. > > > Thanks > Michael > > > > ------------------------------ > > Message: 2 > Date: Tue, 27 Oct 2020 14:13:53 -0700 > From: John Kiniston <johnkinis...@gmail.com> > To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> > Subject: Re: [asterisk-dev] Asterisk 18.0.0 Now Available > Message-ID: > <CAFJQOGfXSK=hMA31AWaB24fZv34Gor4mntPS==v-4mxmlz-...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-dev/attachments/20201027/dd57f47e/attachment.html> > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > --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 > > ------------------------------ > > End of asterisk-dev Digest, Vol 193, Issue 26 > *********************************************
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