The release of Asterisk 16.10.0 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:
New Features made in this release:
-----------------------------------
| [patch] allow Asterisk to set high ToS bits as non-root on Linux | (Reported by Matt Addison) |
Bugs fixed in this release:
-----------------------------------
| Unprotected access to nochecksums variable, causes build failures | (Reported by Guido Falsi) | |
| stream: Enforce formats immutability | (Reported by Joshua C. Colp) | |
| ARI channels cuts the endpoint string over 80 characters | (Reported by sungtae kim) | |
| Crash occurs when fax session switches from T.38 to audio | (Reported by Alexey Vasilyev) | |
| Sporadic crashes with Segmentation fault | (Reported by Joeran Vinzens) | |
| IPv6 addresses in SDP incorrectly formatted | (Reported by Daniel Heckl) | |
| Asterisk REPLY Wrong Contact header port (TCP) | (Reported by Anton Satskiy) | |
| Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used | (Reported by sstream) | |
| AST_MODULE_INFO requires, MODULEINFO does not mention | (Reported by Alexander Traud) | |
| app_confbridge: Add support for disabling text messaging for a user | (Reported by Joshua C. Colp) | |
| pjproject_bundled: Honor --without-pjproject. | (Reported by Alexander Traud) | |
| res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK | (Reported by nappsoft) | |
| chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets | (Reported by Joshua Roys) | |
| res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK | (Reported by nappsoft) | |
| First DTMF is not get | (Reported by Bernard Merindol) | |
| pjsip startup errors when using "with-ssl" configure option | (Reported by Patrick Wakano) | |
| BuildSystem: Search for Python/C API when possibly needed only. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. | (Reported by Alexander Traud) | |
| [patch] chan_sip: TCP/TLS client without server. | (Reported by Alexander Traud) | |
| chan_pjsip: constant DTMF tone if RTP is not setup yet | (Reported by Kevin Harwell) | |
| [patch] bridge_softmix_binaural: Show state in menuselect. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Allow space in path. | (Reported by Alexander Traud) | |
| func_channel: cannot read fields exten, context, userfield, channame from dialplan | (Reported by Sébastien Duthil) | |
| [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. | (Reported by Alexander Traud) | |
| [patch] chan_unistim: Avoid tautological warnings with clang. | (Reported by Alexander Traud) | |
| [patch] test_stasis: Avoid always true warning with clang. | (Reported by Alexander Traud) | |
| res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR | (Reported by Jason Hord) | |
| channel: write to a stream on multi-frame writes | (Reported by Kevin Harwell) | |
| test_utils: incorrectly printing error 'declined to load' | (Reported by Alexander Traud) | |
| func_aes: incorrectly printing error 'declined to load' | (Reported by Alexander Traud) | |
| Crash during conference call using confbridge and video | (Reported by Pascal Cadotte Michaud) | |
| DAHDIRAS fails to properly initiate pppd unless asterisk is running as root | (Reported by Jaco Kroon) | |
| [patch] dundi_read_result crash due to negative number | (Reported by Jaco Kroon) | |
| res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream | (Reported by Joshua C. Colp) | |
| Asterisk is crashing if the 200 OK with SDP | (Reported by sungtae kim) | |
| res_pjsip_session: Allow default non-audio streams to have reflected state | (Reported by Joshua C. Colp) | |
| chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge | (Reported by Michael Neuhauser) | |
| Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. | (Reported by Olivier Krief) | |
| app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup | (Reported by Joshua C. Colp) | |
| Incorrect Sender SSRC in RTCP when p2p rtp bridge is active | (Reported by Torrey Searle) | |
| DTLS Handshake Fails to Occur if ice_support is enabled but not used | (Reported by Torrey Searle) | |
| A non negotiated rtp frame causes call disconnection when there is a SSRC change | (Reported by Paulo Vicentini) | |
| func_enum: ENUM code wrong case | (Reported by Vitold) | |
| Fix the FSF address in the headers of lots of pjproject files | (Reported by Jared Smith) | |
| [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string | (Reported by George Joseph) |
Improvements made in this release:
-----------------------------------
| Missing include on FreeBSD | (Reported by Guido Falsi) | |
| func_volume: Allow decimal numbers as parameter to improve granularity | (Reported by Jean Aunis - Prescom) | |
| dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't | (Reported by Joshua Elson) | |
| Add support for Content-Disposition header in multi-part INVITES | (Reported by Torrey Searle) | |
| res_pjsip_session: Decide more intelligently when to add video | (Reported by Joshua C. Colp) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
Thank you for your continued support of Asterisk!
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