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Revision 1.84 / (download) - annotate - [select for diffs], Wed Apr 19 08:09:20 2023 UTC (7 weeks, 3 days ago) by adam
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Revision 1.83 / (download) - annotate - [select for diffs], Sun Jan 29 21:15:40 2023 UTC (4 months, 1 week ago) by ryoon
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Revision 1.82 / (download) - annotate - [select for diffs], Tue Jan 3 17:36:52 2023 UTC (5 months ago) by wiz
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Revision 1.81 / (download) - annotate - [select for diffs], Wed Nov 23 16:19:31 2022 UTC (6 months, 2 weeks ago) by adam
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Revision 1.79 / (download) - annotate - [select for diffs], Tue Jun 28 11:31:05 2022 UTC (11 months, 1 week ago) by wiz
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Revision 1.78 / (download) - annotate - [select for diffs], Thu May 5 08:20:09 2022 UTC (13 months ago) by nia
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asterisk*: Use OPSYS_VERSION to numerically compare NetBSD versions
Revision 1.77 / (download) - annotate - [select for diffs], Mon Apr 18 19:10:33 2022 UTC (13 months, 3 weeks ago) by adam
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Revision 1.76 / (download) - annotate - [select for diffs], Wed Dec 8 16:03:34 2021 UTC (18 months ago) by adam
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asterisk*: Detect kqueue/timerfd through pkgsrc infrastructure. Fixes PLIST on NetBSD/current.
Revision 1.73 / (download) - annotate - [select for diffs], Wed Sep 29 19:00:24 2021 UTC (20 months, 1 week ago) by adam
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Revision 1.72 / (download) - annotate - [select for diffs], Sun Aug 1 02:57:12 2021 UTC (22 months, 1 week ago) by jnemeth
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asterisk13: Update to Asterisk 13.38.3.
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver
When Asterisk receives a re-INVITE without SDP after having sent
a BYE request a crash will occur. This occurs due to the Asterisk
channel no longer being present while code assumes it is.
* AST-2021-008: Remote crash when using IAX2 channel driver
If the IAX2 channel driver receives a packet that contains an
* AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
handshake
Depending on the timing, it's possible for Asterisk to crash when
using a TLS connection if the underlying socket parent/listener
gets destroyed during the handshake.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
Thank you for your continued support of Asterisk!
Revision 1.71 / (download) - annotate - [select for diffs], Mon May 24 19:49:17 2021 UTC (2 years ago) by wiz
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Revision 1.70 / (download) - annotate - [select for diffs], Wed Apr 21 13:24:28 2021 UTC (2 years, 1 month ago) by adam
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Revision 1.69 / (download) - annotate - [select for diffs], Wed Apr 21 11:41:09 2021 UTC (2 years, 1 month ago) by adam
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Revision 1.68 / (download) - annotate - [select for diffs], Sun Feb 28 22:48:07 2021 UTC (2 years, 3 months ago) by jnemeth
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asterisk13: Update to Asterisk 13.38.2:
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2021-001: Remote crash in res_pjsip_diversion
If a registered user is tricked into dialing a
* AST-2021-002: Remote crash possible when negotiating T.38
When
* AST-2021-003: Remote attacker could prematurely tear down SRTP calls
An unauthenticated remote attacker could replay SRTP packets which could cause
an Asterisk instance configured without strict RTP validation to tear down
calls prematurely.
* AST-2021-004: An unsuspecting user could crash Asterisk with multiple
hold/unhold requests
Due to a signedness comparison mismatch, an authenticated WebRTC client could
cause a stack overflow and Asterisk crash by sending multiple hold/unhold
requests in quick succession.
* AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
Given a scenario where an outgoing call is placed from Asterisk to a remote
SIP server it is possible for a crash to occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
Thank you for your continued support of Asterisk!
Revision 1.67 / (download) - annotate - [select for diffs], Sun Jan 3 09:04:06 2021 UTC (2 years, 5 months ago) by jnemeth
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Disable -march=native default.
Revision 1.66 / (download) - annotate - [select for diffs], Sat Jan 2 22:45:43 2021 UTC (2 years, 5 months ago) by jnemeth
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Update to Asterisk 13.38.1
-----
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18. The available releases are released as versions
13.38.1, 16.15.1, 17.9.1 and 18.1.1.
The following security vulnerabilities were resolved in these versions:
* AST-2020-003: Remote crash in res_pjsip_diversion
A crash can occur in Asterisk when a SIP message is received that has a
History-Info header, which contains a tel-uri.
* AST-2020-004: Remote crash in res_pjsip_diversion
A crash can occur in Asterisk when a SIP 181 response is received that has a
Diversion header, which contains a tel-uri.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.38.0.
The release of Asterisk 13.38.0 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-29057 - pjsip: Crash on call rejection during high load
(Reported by Sandro Gauci)
Improvements made in this release:
-----------------------------------
* ASTERISK-29056 - Increase reg_server column size for
ps_contacts table realtime
(Reported by sungtae kim)
* ASTERISK-29055 - Create a Bridge with video_single mode
(Reported by sungtae kim)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
invites (with auth) on 407 replies
(Reported by Sebastian Damm)
* ASTERISK-29108 - resource_endpoints.c : Memory leak if
endpoint not found
(Reported by Jean Aunis - Prescom)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName
(Reported by Eric Smith)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported by Sebastian Damm)
* ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by ?????????)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-29081 - res_stasis: Add compare function for bridges
moh container
(Reported by Hajek Michal)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P??ter Juh??sz)
* ASTERISK-28416 - Unable to get rtp codec payload code for slin
(Reported by Brian J. Murrell)
New Features made in this release:
-----------------------------------
* ASTERISK-29027 - Implement support for History-Info
(Reported by Torrey Searle)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.38.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk
16.8. The available releases are released as versions 13.37.1,
16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
The following security vulnerabilities were resolved in these versions:
* AST-2020-001: Remote crash in res_pjsip_session
Upon receiving a new SIP Invite, Asterisk did not return the created dialog
locked or referenced.
* AST-2020-002: Outbound INVITE loop on challenge with different nonce.
If Asterisk is challenged on an outbound INVITE and the nonce is changed in
each response, Asterisk will continually send INVITEs in a loop. This causes
Asterisk to consume more and more memory since the transaction will never
terminate (even if the call is hung up), ultimately leading to a restart or
shutdown of Asterisk. Outbound authentication must be configured on the
endpoint for this to occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.37.0.
The release of Asterisk 13.37.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
* ASTERISK-29043 - app_queue: Leave empty sometimes not
recorded as abandoned
(Reported by Kfir Itzhak)
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied
(Reported by Misha Vodsedalek)
* ASTERISK-29029 - Voicemail "pollmailboxes"-option not
working, bug in function handle_subscribe
(Reported by Karsten Wemheuer)
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
* 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 - res_speech: Assertion on format
(Reported by Nickolay V. Shmyrev)
* ASTERISK-29001 - chan_pjsip does not process or forward 181 responses
(Reported by Torrey Searle)
* 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 - res_pjsip_session: Aggressively terminates
session on failed re-INVITE
(Reported by Joshua C. Colp)
* ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
appended RTP string to each message block.
(Reported by Thomas Johnson)
Improvements made in this release:
-----------------------------------
* ASTERISK-29010 - Allow disabling of FollowMe prompt
(Reported by Dennis)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.37.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.36.0.
The release of Asterisk 13.36.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied
(Reported by Misha Vodsedalek)
* ASTERISK-29029 - Voicemail "pollmailboxes"-option not
working, bug in function handle_subscribe
(Reported by Karsten Wemheuer)
* 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-29011 - chan_sip: ToHost property not cleared on reload
(Reported by Dennis)
* ASTERISK-28987 - BridgeCreated ARI event shows wrong
video_mode info
(Reported by sungtae kim)
* ASTERISK-28927 - Asterisk crash in music on hold
(Reported by David Cunningham)
* 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 - res_pjsip_registrar: Expires on statically
configured contacts is not correct
(Reported by tootai)
* ASTERISK-28978 - acl: named_acl rule misconfiguration results
in segfault on reading rule from realtime
(Reported by Andrew Yager)
* ASTERISK-28975 - res_http_websocket: Text payload data
doesn't necessary include trailing zero
(Reported by Nickolay V. Shmyrev)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.36.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.35.0.
The release of Asterisk 13.35.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
* ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
contacts on AOR
(Reported by Joshua C. Colp)
* ASTERISK-28930 - ./configure --without-ssl build failure
(Reported by Jaco Kroon)
* ASTERISK-28957 - chan_sip: chan_sip does not process 400
response to an INVITE.
(Reported by Frederic LE FOLL)
* ASTERISK-28888 - res_corosync: causes asterisk crash in huge
distributed environment.
(Reported by Universit?? di Bologna - CESIA VoIP)
* ASTERISK-28955 - "setvar" doesn't work properly in
dahdi-channels.conf
(Reported by Marin Odrljin)
* ASTERISK-28942 - res_sorcery_memory_cache: Individual object
expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
* ASTERISK-28952 - Queue wrapuptime sometimes not respected
(based on stale lastcall time)
(Reported by Walter Doekes)
* ASTERISK-28950 - Stale code in app_queue to check untouched channel
(Reported by Walter Doekes)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
* ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
* ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
* ASTERISK-28900 - res_fax: Double frame free when gateway in
use with off-nominal format usage
(Reported by Gregory Massel)
* ASTERISK-28929 - pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28885 - res_rtp_asterisk: Simultaneous termination
and ICE complete can cause crash
(Reported by Josep B)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
* ASTERISK-28959 - res_pjsip: Added option for disable rport
parameter set
(Reported by sungtae kim)
* ASTERISK-28958 - Continue reading string when ping received
by websocket
(Reported by Nickolay V. Shmyrev)
* ASTERISK-28945 - AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
* ASTERISK-28949 - res_http_websocket: Add masking to websocket client
(Reported by Moises Silva)
* ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.35.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.34.0.
The release of Asterisk 13.34.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
* ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing
(Reported by nappsoft)
* ASTERISK-28884 - x-ast-orig-host not filtered out from
request URI and To header
(Reported by nappsoft)
* ASTERISK-28898 - bridge_softmix: Conference bridge not
passing silent rtp packets
(Reported by Jonathan Hunter)
* ASTERISK-28904 - RTP ICE leaks the memory
(Reported by sungtae kim)
* ASTERISK-28854 - SIGSEGV when pjsip show history encounters
IPV6 address
(Reported by Roger James)
* ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
truncation.
(Reported by Alexander Traud)
* ASTERISK-28776 - Non async-signal-safe syscalls used after
fork before exec
(Reported by nappsoft)
* ASTERISK-28829 - app_queue: leaking stasis subscription when
Redirecting call
(Reported by lvl)
* ASTERISK-25844 - app_queue: Ghost channels in "core show
channels" output
(Reported by Etienne Lessard)
* ASTERISK-22920 - Crash while Forwarding from TLS extension
with CHANNEL args secure_bridge_media and
secure_bridge_signaling
(Reported by Shlomi Gutman)
* ASTERISK-28859 - pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
* ASTERISK-28852 - Unprotected access to nochecksums variable,
causes build failures
(Reported by Guido Falsi)
Improvements made in this release:
-----------------------------------
* ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
(Reported by Joshua C. Colp)
* ASTERISK-28879 - pjproject has race conditions in it's build system
(Reported by Guido Falsi)
* ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms
(Reported by Guido Falsi)
* ASTERISK-28832 - chan_mobile creates PCMA streams that make
some VoIP clients crash or not render received audio
(Reported by Peter Turczak)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.34.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.33.0.
The release of Asterisk 13.33.0 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-28813 - func_volume: Allow decimal numbers as
parameter to improve granularity
(Reported by Jean Aunis - Prescom)
* ASTERISK-27946 - dial (API): Storage of dialed target uses
AST_MAX_EXTENSION when it shouldn't
(Reported by Joshua Elson)
* ASTERISK-28782 - Add support for Content-Disposition header
in multi-part INVITES
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28852 - Unprotected access to nochecksums variable,
causes build failures
(Reported by Guido Falsi)
* ASTERISK-28847 - ARI channels cuts the endpoint string over
80 characters
(Reported by sungtae kim)
* ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted
(Reported by Daniel Heckl)
* ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP)
(Reported by Anton Satskiy)
* 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 - AST_MODULE_INFO requires, MODULEINFO does
not mention
(Reported by Alexander Traud)
* ASTERISK-28837 - pjproject_bundled: Honor
--without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
ignoring TCP and TLS sockets
(Reported by Joshua Roys)
* ASTERISK-28812 - First DTMF is not get
(Reported by Bernard Merindol)
* ASTERISK-28758 - pjsip startup errors when using "with-ssl"
configure option
(Reported by Patrick Wakano)
* ASTERISK-28824 - BuildSystem: Search for Python/C API when
possibly needed only.
(Reported by Alexander Traud)
* ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
Programming Language is python-2.7.
(Reported by Alexander Traud)
* ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
* ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
setup yet
(Reported by Kevin Harwell)
* ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
doc/pdf leftovers.
(Reported by Alexander Traud)
* ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
(Reported by Alexander Traud)
* ASTERISK-28801 - [patch] stasis: Avoid always true warnings
with clang.
(Reported by Alexander Traud)
* ASTERISK-28796 - func_channel: cannot read fields exten,
context, userfield, channame from dialplan
(Reported by S??bastien Duthil)
* ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-28808 - [patch] test_stasis: Avoid always true
warning with clang.
(Reported by Alexander Traud)
* ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
endpoint synchronization for a specific AOR
(Reported by Jason Hord)
* ASTERISK-28789 - test_utils: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-28788 - func_aes: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
unless asterisk is running as root
(Reported by Jaco Kroon)
* ASTERISK-21205 - [patch] dundi_read_result crash due to
negative number
(Reported by Jaco Kroon)
* ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
* ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
triggered during direct-media (native_rtp) bridge
(Reported by Michael Neuhauser)
* ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
are not consistent with examples. Missing examples.
(Reported by Olivier Krief)
* ASTERISK-28780 - app_mixmonitor: Memory leak due to race
condition between AMI MixMonitor and hangup
(Reported by Joshua C. Colp)
* ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
bridge is active
(Reported by Torrey Searle)
* ASTERISK-28759 - A non negotiated rtp frame causes call
disconnection when there is a SSRC change
(Reported by Paulo Vicentini)
* ASTERISK-26711 - func_enum: ENUM code wrong case
(Reported by Vitold)
* ASTERISK-23407 - Fix the FSF address in the headers of lots
of pjproject files
(Reported by Jared Smith)
* ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
is enabled but not used
(Reported by Torrey Searle)
* ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
makes DNS calls and always returns an empty string
(Reported by George Joseph)
New Features made in this release:
-----------------------------------
* ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
as non-root on Linux
(Reported by Matt Addison)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.33.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.32.0.
The release of Asterisk 13.32.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28766 - PJSIP blind transfer not completed after
using Proceeding()
(Reported by lvl)
* ASTERISK-28685 - check_expr2: linking (when hardening) and
cross-compiling troubles
(Reported by Sebastian Kemper)
* ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
the "variables" field
(Reported by Jean Aunis - Prescom)
* ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold
(Reported by Ross Beer)
* ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
complete before allowing sending
(Reported by Benjamin Keith Ford)
* ASTERISK-28697 - res_pjsip: Named ACL does not update on
reload if changed
(Reported by Timothy Vanderaerden)
* ASTERISK-28738 - Incorrect state machine used when
MOH_PASSTHRU is used
(Reported by Torrey Searle)
* ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
to SLIN
(Reported by Ross Beer)
* ASTERISK-26955 - pjsip: SIP Packets with Via "received="
Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)
* ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
depleted, should return 503
(Reported by Walter Doekes)
* ASTERISK-28719 - Cannot remove defaultrule from queue using
realtime queues
(Reported by EDV O-TON)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
* ASTERISK-26082 - res_pjsip_messaging: MessageSend
Content-Type can't be changed
(Reported by Alex)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28679 - stasis application is destroyed after its creation
(Reported by Francois Blackburn)
* ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in
spite of the error when sending
(Reported by Dmitriy Serov)
* ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
Asterisk To Drop Calls
(Reported by Paul Brooks)
* ASTERISK-28686 - chan_sip strictrtp=yes fails when media
source is changed: no audio
(Reported by Walter Doekes)
Improvements made in this release:
-----------------------------------
* ASTERISK-28750 - TLS/SSL Key too small error
(Reported by Martin Zeh)
* ASTERISK-24798 - Documentation - Clarify That Format Is Set
By File Name Extension In MixMonitor
(Reported by xrobau)
* ASTERISK-28726 - install_prereq script uses the interactive
mode when installing aptitude
(Reported by Sylvain Afchain)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.32.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.31.0.
The release of Asterisk 13.31.0 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:
-----------------------------------
* ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
/ "maxredirs" doesn't do anything
(Reported by candrews)
* ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
ability to match on source port
(Reported by Sean Bright)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28679 - stasis application is destroyed after its creation
(Reported by Francois Blackburn)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
* ASTERISK-28677 - CDR billsec is always 0 for transferred calls
(Reported by Maciej Michno)
* ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
translation' output
(Reported by Sean Bright)
* ASTERISK-24484 - Update documentation for statsd module -
usage requirements unclear
(Reported by Dan Jenkins)
* ASTERISK-28702 - chan_dahdi: holding a channel via flash to
dialtone times out after 0:16:40
(Reported by Andrew Siplas)
* ASTERISK-28695 - core: minmemfree watermark uses free RAM,
not available RAM
(Reported by Kevin Flyn)
* ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
whitespace appears empty in the dialplan
(Reported by Frank Matano)
* ASTERISK-23739 - [patch]Segfault forwarding voicemail with
ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
* ASTERISK-27622 - empty voicemail.conf required for ARA
(realtime) voicemail to leave message
(Reported by Jim Van Meggelen)
* ASTERISK-28349 - Pause reason not reported in QueueMember AMI event
(Reported by Niksa Baldun)
* ASTERISK-21794 - CLI command 'realtime update2' syntax
failure when using according to usage help
(Reported by Cedric BASSAGET)
* ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
support for hostnames
(Reported by Joshua C. Colp)
* ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
be present instead of just one
(Reported by AvayaXAsterisk)
* ASTERISK-28682 - app_record: Lack of `beep` audio file causes
application to return error and hangup
(Reported by Corey Farrell)
* ASTERISK-28507 - Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
* ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
does not preserve XML <dialog-info> version number
(Reported by Bryan Nelson)
* ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
with concurrent command pri show span X
(Reported by Dirk Wendland)
* ASTERISK-28633 - stasis bridge topic leak
(Reported by Joeran Vinzens)
* ASTERISK-28492 - pjsip reload not reloading wizard
endpoint/pickup_group endpoint/call_group
(Reported by Jean-Denis Girard)
* ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
it's supposed to due to invalid syntax
(Reported by Richard Kenner)
* ASTERISK-28497 - func_odbc: truncating Unicode string on readsql
(Reported by Boris P. Korzun)
* ASTERISK-28647 - chan_sip: RTP frames not transmitted after
emitting a COLP
(Reported by Jean Aunis - Prescom)
* ASTERISK-28667 - Asterisk ignores parsing of config files if
a Byte order mark is present
(Reported by Robin Leffmann)
* ASTERISK-28664 - "trustrpid" is misspelled in
sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28663 - jansson: Support old versions
(Reported by Joshua C. Colp)
* ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
fails to deactivate CDR.
(Reported by Frederic LE FOLL)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
with config option
(Reported by Kevin Harwell)
* ASTERISK-28628 - Debian 10.2: Warning when app_voicemail is compiling
(Reported by Stanislav Abramenkov)
* ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
documentation
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28651 - chan_sip logs errors on tx to non-existent
TCP connections
(Reported by Jaco Kroon)
* ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
200 Response Contact
(Reported by Ross Beer)
Improvements made in this release:
-----------------------------------
* ASTERISK-28710 - Should be able to disable the /httpstatus
URI in the built-in HTTP server
(Reported by Sean Bright)
* ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail
(Reported by cmaj)
* ASTERISK-28673 - GET FULL VARIABLE documentation clarification
(Reported by Jonathan Harris)
* ASTERISK-28658 - app_confbridge: Add support for setting
maximum sample rate
(Reported by Joshua C. Colp)
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.30.0.
The release of Asterisk 13.30.0 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 - chan_sip: Depending on configuration an
INVITE can alter Addr of a peer
(Reported by Andrey V. T.)
* ASTERISK-28580 - Bypass SYSTEM write permission in manager
action allows system commands execution
(Reported by Eliel Sarda½½ons)
Improvements made in this release:
-----------------------------------
* ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
retries reached
(Reported by Daniel)
* ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
(Reported by Sam Banks)
* ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
backend when format differs from attachfmt column
(Reported by cmaj)
* 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 - [patch] add the ability for asterisk to
generate on-hold re-invites
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28663 - jansson: Support old versions
(Reported by Joshua C. Colp)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28641 - res_pjsip Segfaults when realtime
configuration to an AOR points to a not existent AOR
(Reported by Ross Beer)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
* 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 - res_pjsip_session: ast_json_vpack: Invalid
UTF-8 string on hangup when TEST_FRAMEWORK enabled
(Reported by Bernhard Schmidt)
* ASTERISK-28631 - res_parking: Doesn't park when parkee and
parker are the same
(Reported by Ross Beer)
* ASTERISK-28612 - res_pjsip_t38: crash on reinvite with zero
port and no c= line
(Reported by Salah Ahmed)
* ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok received
(Reported by Salah Ahmed)
* ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
Active" after a short alarm
(Reported by Frederic LE FOLL)
* ASTERISK-28616 - parking: Deadlock when multi call parking
(Reported by Joshua C. Colp)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28608 - app_amd: Use time calculation to calculate timeout
(Reported by Michael Cargile)
* ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
sent packet length doesn't match
(Reported by Joshua Elson)
* ASTERISK-28618 - bridge_softmix: hold not cleared when
joining a softmix bridge
(Reported by Kevin Harwell)
* ASTERISK-26481 - FILE function grabs garbage along with read
data when target line has no newline
(Reported by Jonathan Harris)
* ASTERISK-28572 - Memory leaks in res_calendar_exchange and
res_calendar_icalendar
(Reported by Yoooooo Ha)
* ASTERISK-28585 - ari/resource_events: Crash in event session cleanup
(Reported by Kevin Harwell)
* ASTERISK-28590 - utils.c throws repeated warnings;
"pthread_attr_setstacksize: Invalid argument"
(Reported by Speed Dial Dave)
* ASTERISK-28578 - race condition on pjsip channelstats command
(Reported by Salah Ahmed)
* ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
removed) column
(Reported by Christoph Moench-Tegeder)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5
(Reported by Niklas Larsson)
* ASTERISK-28561 - Asterisk Deadlocks
(Reported by Aheliotech)
* ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI
(Reported by Jeremiah Gadd)
* ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
unsolicited_mwi container
(Reported by Kevin Harwell)
* ASTERISK-28566 - CDR backend unload problem during active call(s)
(Reported by Marian Piater)
* ASTERISK-28544 - Wrong contact representation in ipv6 mode
(Reported by J½½rgen H)
* ASTERISK-28534 - Segmentation fault when there is no priority
for an extension
(Reported by Timothy Vanderaerden)
* ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
is configured
(Reported by Juan Martin)
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported by Cyril Rami½½re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD
(Reported by Guido Falsi)
* ASTERISK-23756 - setvar directive when used in template and a
child of said template, results in duplicate variable names
(Reported by Michael Goryainov)
New Features made in this release:
-----------------------------------
* ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
PlayDTMF instead of only "sending"
(Reported by lvl)
* ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header
(Reported by Martin Tomec)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.30.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 16 and 17, and Certified Asterisk 13.21.
The available releases are released as versions 13.29.2, 16.6.2,
17.0.1 and 13.21-cert5.
The following security vulnerabilities were resolved in these versions:
* AST-2019-006: SIP request can change address of a SIP peer.
A SIP request can be sent to Asterisk that can change a SIP peer½½½ó IP
address. A REGISTER does not need to occur, and calls can be hijacked as a
result. The only thing that needs to be known is the peer½½½ó name;
authentication details such as passwords do not need to be known. This
vulnerability is only exploitable when the ½½½îat½½option is set to the
default, or ½½½áuto_force_rport½½
* AST-2019-007: AMI user could execute system commands.
A remote authenticated Asterisk Manager Interface (AMI) user without
½½½óystem½½authorization could use a specially crafted ½½½Ïriginate½½AMI
request to execute arbitrary system commands.
* AST-2019-008: Re-invite with T.38 and malformed SDP causes crash.
If Asterisk receives a re-invite initiating T.38 faxing and has a port of 0
and no c line in the SDP, a crash will occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.29.1.
The release of Asterisk 13.29.1 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5
(Reported by Niklas Larsson)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.1
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.29.0.
The release of Asterisk 13.29.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported by Cyril Rami½½re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD
(Reported by Guido Falsi)
* ASTERISK-28527 - ChanIsAvail() creates a CDR if
unanswered=yes is set in cdr.conf
(Reported by Frederic LE FOLL)
* ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
PRI channel hangs up
(Reported by Frederic LE FOLL)
* ASTERISK-28511 - codec_resample: Bad sound quality when up
sampling from SLIN16 to SLIN32
(Reported by Ruddy G)
* ASTERISK-28499 - translate: Crash when frame does not have a
"src" field set
(Reported by Gregory Massel)
* ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
type not at end of a struct
(Reported by Alexander Traud)
* ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register
(Reported by Chris Savinovich)
* ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
characters, NEC only supports up to 32 characters
(Reported by Dan Cropp)
* ASTERISK-28505 - app_voicemail/IMAP: segfault in
leave_voicemail because not checking mailstream
(Reported by Alexei Gradinari)
* ASTERISK-28487 - compile menuselect on gentoo
(Reported by Kilburn)
* ASTERISK-28472 - Asterisk occasionally passes a NULL as
srtp->session to srtp_protect/unprotect causing SEGV
(Reported by Jonas Swiatek)
* ASTERISK-28498 - cel / cdr: Event times may be incorrect
(Reported by Joshua C. Colp)
* ASTERISK-28483 - packet lost on UDPTL wrap around
(Reported by Torrey Searle)
* ASTERISK-28480 - json integer overflow in ssrc and timestamp
(Reported by Salah Ahmed)
* ASTERISK-28228 - res_pjsip: pjsip show contacts prints double entries
(Reported by Ian Jones)
* ASTERISK-28477 - Crash when not specifying "dbfile" in
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28478 - Crash performing "core reload" with modified
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
deadlocks (in chan_sip)
(Reported by Walter Doekes)
New Features made in this release:
-----------------------------------
* ASTERISK-17808 - [patch] Unregister a realtime moh class
(Reported by Byron Clark)
* ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
chan_pjsip to setup From header URI domain
(Reported by Stas Kobzar)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16. The available releases are
released as versions 13.28.1, 15.7.4 and 16.5.1.
The following security vulnerabilities were resolved in these versions:
* AST-2019-004: Crash when negotiating for T.38 with a declined stream
When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
responds with a declined media stream a crash will then occur in Asterisk.
* AST-2019-005: Remote Crash Vulnerability in audio transcoding
When audio frames are given to the audio transcoding support in Asterisk the
number of samples are examined and as part of this a message is output to
indicate that no samples are present. A change was done to suppress this
message for a particular scenario in which the message was not relevant. This
change assumed that information about the origin of a frame will always exist
when in reality it may not.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.28.0
The release of Asterisk 13.28.0 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-28447 - res_pjsip_messaging: In-dialog MESSAGE with
no body causes crash
(Reported by Gil Richard)
* ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 reINVITE
(Reported by Francesco Castellano)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
systems caused by ASTERISK-28317
(Reported by abelbeck)
* ASTERISK-26006 - Show offending IP for TLS setup failures in logs
(Reported by Oleksandr Natalenko)
* ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged
(Reported by Bernhard Schmidt)
* ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak
with specific usage
(Reported by Joshua C. Colp)
* ASTERISK-28018 - IP Fragmentation happening instead of DTLS
fragmentation on handshake server hello certificate
(Reported by vijay kumar)
* ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
Asterisk attempts to generate hangup event
(Reported by Abhay Gupta)
* ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
(Reported by Dmitry Svyatogorov)
* ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying
(Reported by pasandev)
* ASTERISK-28419 - app_amd: Does not work with silence suppression
(Reported by Nasir Iqbal)
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido Falsi)
* ASTERISK-27994 - PJSIP: Early media ringback not indicated
after Progress()
(Reported by Gregory Massel)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21.
The available releases are released as versions 13.27.1, 15.7.3,
16.4.1 and 13.21-cert4.
The following security vulnerabilities were resolved in these versions:
* AST-2019-002: Remote crash vulnerability with MESSAGE messages
A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash.
* AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an
endpoint to switch it to T.38. If the endpoint responds with an improperly
formatted SDP answer including both a T.38 UDPTL stream and an audio or video
stream containing only codecs not allowed on the SIP peer or user a crash will
occur. The code incorrectly assumes that there will be at least one common
codec when T.38 is also in the SDP answer.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.27.0.
The release of Asterisk 13.27.0 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:
-----------------------------------
* ASTERISK-28375 - res_pjsip: New configuration setting to
allow disabling norefersub
(Reported by Dan Cropp)
* ASTERISK-28320 - Added ARI resource
/ari/channels/{channelid}/rtp_statistics
(Reported by sungtae kim)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido Falsi)
* ASTERISK-28412 - GCC 9 catches more string formatting issues
(Reported by George Joseph)
* ASTERISK-28392 - The no-partial-inlining flag isn't passed to
the bundled pjproject or jansson builds
(Reported by George Joseph)
* ASTERISK-28402 - res_pjsip_registrar: SEGV in
registrar_find_contact
(Reported by Ross Beer)
* ASTERISK-28143 - app_amd: Infinite loop on silent calls
(Reported by Abhay Gupta)
* ASTERISK-28353 - stasis: Crash at shutdown when statistics enabled
(Reported by Joshua C. Colp)
* ASTERISK-28374 - latest asterisk unconditionally launch gcc
--version, even if the compiler is different
(Reported by Guido Falsi)
* ASTERISK-28391 - res_indications: Crash requesting
autocomplete on indications cli command
(Reported by Lucas Mendes)
* ASTERISK-27935 - app_voicemail: emailbody per user can't
contain commas
(Reported by S½½bastien Duthil)
* ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
extensions with '-' in them
(Reported by test011)
* ASTERISK-17799 - AEL reload causes loss of control in a macro
(Reported by Kirill Katsnelson)
* ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter
(Reported by Luke-Jr)
* ASTERISK-14939 - AEL parsers does not find existing label
(Reported by klaus3000)
* ASTERISK-20182 - Parsing a label beginning with a numeric
character in all Goto/GotoIf/GotoIfTime application causes
unexpected behavior
(Reported by Janu)
* ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323 Disabled
(Reported by Dmitry Shubin)
* ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
lead to both inband and info
(Reported by Salah Ahmed)
* ASTERISK-28362 - strtok_r() makes gcc compile warning
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28363 - Millisecond-resolution call stats including
PDD in channel variables
(Reported by Antoni Goldstein)
* ASTERISK-20207 - Asterisk should clear out any .lock files in
the voice mail directory on startup.
(Reported by Steven Wheeler)
* ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
work with.
(Reported by Corey Farrell)
* ASTERISK-28343 - Added app_name, app_data to channel type
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.27.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.26.0.
The release of Asterisk 13.26.0 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:
-----------------------------------
* ASTERISK-28267 - res_stasis: Add ability to switch
applications
(Reported by Benjamin Keith Ford)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
(Reported by Olivier Krief)
* ASTERISK-28350 - manager: Stasis backed up due to locking
(Reported by Joshua C. Colp)
* ASTERISK-25792 - chan_sip: qualifygap bounds checking
(Reported by Paul Sandys)
* ASTERISK-28341 - res_config_odbc eliminates empty custom (½½½À½½ prefix) variables
(Reported by Alexei Gradinari)
* ASTERISK-28333 - StasisEnd event makes wrong timestamp value
(Reported by sungtae kim)
* ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
minutes to be sent
(Reported by Jared Hull)
* ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
without channel lock or reference
(Reported by Francisco Seratti)
* ASTERISK-28314 - ARI: API changed but "apiVersion" in
rest-api\resources.json did not
(Reported by Stefan Repke)
* ASTERISK-28335 - stasis: Make topic and maybe subscription
names unique and more useful
(Reported by Joshua C. Colp)
* ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
zero for rtcp stat calculation
(Reported by sungtae kim)
* ASTERISK-28332 - Variable ALTCONF ignored when service is
used in Debian
(Reported by Cirillo Ferreira)
* ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
183 without SDP
(Reported by Torrey Searle)
* ASTERISK-28328 - MeetMe global non-admin mute is muting
admins that subsequently join
(Reported by Philip Mott)
* ASTERISK-28168 - app_queue: Adding a blank entry into sql
queue_members crashes asterisk.
(Reported by Michael)
* ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion script fails
(Reported by Guido Weckwerth)
* ASTERISK-28272 - The basic-pbx config samples don't produce a
running asterisk
(Reported by George Joseph)
* ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
field after handling a 302 redirect
(Reported by Alex Odrov)
* ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
license header
(Reported by Jeremy Lain½½)
* ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
changing voicemail password with ODBC
(Reported by Michael)
* ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
multiple UDP interfaces
(Reported by Nikolay shakin)
* ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
pjsip_wizard.conf causes crash
(Reported by Jonathan Harris)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28301 - Allow voicemail boxes to be subscribed to
with a presence event package
(Reported by George Joseph)
* ASTERISK-28303 - res_rtp_asterisk: Interaction between
smoother and DTMF can cause out of order timestamps
(Reported by Torrey Searle)
* ASTERISK-28302 - ARI: "Error destroying mutex" when listing
all ARI applications
(Reported by Stefan Repke)
* ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
applications
(Reported by George Joseph)
* ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
events when GETting causes overload of events
(Reported by George Joseph)
* ASTERISK-28284 - switching between native_bridge and
simple_bridge can cause one way audio
(Reported by Torrey Searle)
* ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls
(Reported by Paulo Vicentini)
* ASTERISK-28251 - CI: Fix CI so it reverifies commit message changes
(Reported by George Joseph)
* ASTERISK-28277 - database: Add some basic logging
(Reported by Joshua C. Colp)
* ASTERISK-28181 - ari: Originating overwrites channel start time
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28326 - ari: Added timestamp for some ari events.
(Reported by sungtae kim)
* ASTERISK-28317 - Add logical group at DAHDIChannel event and
create "dahdi_group" at CHANNEL function
(Reported by Cirillo Ferreira)
* ASTERISK-28279 - Added creation timestamp for bridge
(Reported by sungtae kim)
* ASTERISK-28292 - Changed to show all channel stats including
wrong media
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.26.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.25.0.
The release of Asterisk 13.25.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls
(Reported by Paulo Vicentini)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile
(Reported by David Wilcox)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1
line dialplan context for stasis apps
(Reported by George Joseph)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI
(Reported by Ray)
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe
request during res_parking reload
(Reported by Giuseppe Sucameli)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate
and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28250 - build: Cross-compilation fails for target
arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28156 - Race condition involving session->media
(res_pjsip_session) leads to crash.
(Reported by Paulo Vicentini)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
break data reception
(Reported by Jeremy Lain½½)
* ASTERISK-28252 - HangupHandler manager events are never thrown
(Reported by Gerald Schnabel)
* ASTERISK-28231 - res_http_websocket: Not responding to
Connection Close Frame (opcode 8)
(Reported by Jeremy Lain½½)
* ASTERISK-28249 - res_monitor: Segfault with
Monitor(wav,file,i)
(Reported by Valentin Vidi½½)
* ASTERISK-28244 - stasis: Filter messages at publishing to AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the
channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28232 - core: RAII using clang use-after-scope issue
(Reported by Diederik de Groot)
* ASTERISK-28225 - app_voicemail: Channel variable
VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by Andrew Nagy)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on RTP renegotiation
(Reported by Alexei Gradinari)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
negotiation problem
(Reported by David Kuehling)
* ASTERISK-28117 - stasis: Add statistics for usage when in
developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28201 - [patch] confbridge: no announce to the
marked users when they join an empty conference
(Reported by Alexei Gradinari)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-28186 - stasis: Filter messages at publishing based
on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is
set to invite, we're not trying UPDATE
(Reported by George Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is
set to invite, asterisk is not trying UPDATE
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
* ASTERISK-28246 - Support skipping on the g726 format
(Reported by Eyal Hasson)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.24.1.
The release of Asterisk 13.24.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.1
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.24.0.
The release of Asterisk 13.24.0 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-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
New Features made in this release:
-----------------------------------
* ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
in Contact header in chan_pjsip
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28125 - app_queue: Revert broken queue channel
reference patch
(Reported by lvl)
* ASTERISK-28151 - app_voicemail: MWI fails with
mailboxes=##@device instead of mailboxes=##@default
(Reported by Ronald Raikes)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_* modules unload
(Reported by sungtae kim)
* ASTERISK-28159 - SIGABRT caused by stack corruption in
hashkeys_read when no matching keys present
(Reported by Michael Walton)
* ASTERISK-28140 - repeated segmentation faults
(Reported by Eyal Hasson)
* ASTERISK-28103 - stasis: Filter messages at publishing to
reduce work done
(Reported by Joshua C. Colp)
* ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
Re-Invite omits routset
(Reported by Torrey Searle)
* ASTERISK-28158 - Some conditions prevent running of el_end,
break the terminal.
(Reported by Corey Farrell)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on voice packet with marker bit
(Reported by Alexei Gradinari)
* ASTERISK-28110 - rtp: Incorrect Packetization
(Reported by Robert Cripps)
* ASTERISK-28146 - pbx_config: Only the first [globals] section
is processed.
(Reported by Corey Farrell)
* ASTERISK-28150 - Formatting error in documentation
(Reported by Scott Griepentrog)
* ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
report AST_CEL_PICKUP in handle_invite_replaces
(Reported by Luit van Drongelen)
* ASTERISK-28137 - res_pjsip_notify: improve realtime
performance on CLI completion on the endpoint
(Reported by Alexei Gradinari)
* ASTERISK-27980 - Caller ID cannot be changed on Attended
Transfer before dialing out
(Reported by Alexei Gradinari)
* ASTERISK-28089 - function ast_sendtext() create RTP realtime
packets with a trailing null byte in the payload
(Reported by Emmanuel BUU)
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
(Reported by Emmanuel BUU)
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
(Reported by Cao Minh Hiep)
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
(Reported by Joshua C. Colp)
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI
(Reported by Peter Katzmann)
* ASTERISK-28049 - res_pjproject build failure
(Reported by Jaco Kroon)
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
* ASTERISK-28032 - Realtime queuemembers are not updated during retry phase
(Reported by lvl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not boolean
(Reported by Joshua C. Colp)
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
(Reported by Sean Bright)
Improvements made in this release:
-----------------------------------
* ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
parse an URI and return a specified part of the URI
(Reported by Alexei Gradinari)
* ASTERISK-28136 - Allow the sip_to_pjsip script to be used in a pipe
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28046 - Remove stale nonoptreq references
(Reported by Walter Doekes)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.21.
The available releases are released as versions 13.23.1, 14.7.8,
15.6.1 and 13.21-cert3.
These releases are available for immediate download at
The following security vulnerabilities were resolved in these versions:
* AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade
There is a stack overflow vulnerability in the res_http_websocket.so module of
Asterisk that allows an attacker to crash Asterisk via a specially crafted
HTTP request to upgrade the connection to a websocket. The attacker½½½ó
request causes Asterisk to run out of stack space and crash.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.23.0.
The release of Asterisk 13.23.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
authentification error
(Reported by Ian Gilmour)
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
(Reported by Florian Floimair)
* ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock
mutex 'peer' without owning it!
(Reported by Alec Davis)
* ASTERISK-28002 - When T.140 realtime text is negociated, a
lot of debug traces are generated
(Reported by Emmanuel BUU)
* ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
LEAVEEMPTY
(Reported by Valentin Safonov)
* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer
(Reported by Torrey Searle)
* ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
Do not undef s_addr.
(Reported by Alexander Traud)
* ASTERISK-28001 - res_pjsip_registrar: Improve performance of
inbound handling
(Reported by Joshua Colp)
* ASTERISK-27999 - Wrong SRTP use status report
(Reported by Salah Ahmed)
* ASTERISK-27966 - pjsip: Race condition in 183 re transmission
can result in a deadlock
(Reported by Torrey Searle)
* ASTERISK-15331 - make menuselect fails due to undefined
symbols (initscr32, w32addch) in menuselect_curses.o
(Reported by Majdi Bsoul)
* ASTERISK-14935 - [regression] menuselect compilation failure on Solaris 10
(Reported by Samuel Owens)
* ASTERISK-12382 - menuselect compilation failure on Solaris 10 / gcc 3.4.3
(Reported by rleasure)
* ASTERISK-9107 - menuselect compilation failure on Solaris 10/gcc-4.1.1
(Reported by Bob Atkins)
* ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
matches against "generic string" headers
(Reported by George Joseph)
* ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
(Reported by Alexander Traud)
* ASTERISK-27591 - Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua Colp)
* ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
if not in offer
(Reported by Torrey Searle)
* ASTERISK-27880 - [patch] pjproject_bundled: Repair
./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua Colp)
* ASTERISK-27972 - res_sorcery_config: Allow object name based matching
(Reported by Joshua Colp)
* ASTERISK-27967 - srtp: rejecting short sdes lifetimes
incompatible with obihai ATAs
(Reported by Nick French)
* ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
printing headers in sip_msg
(Reported by Nick French)
* ASTERISK-27563 - pjsip modules always get -O2 even when
DONT_OPTIMIZE is set
(Reported by George Joseph)
* ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
(Reported by Alexander Traud)
* ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
(Reported by Alexander Traud)
* ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire
(Reported by Alexei Gradinari)
* ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
correctly handle two Reason headers
(Reported by Ross Beer)
* ASTERISK-27763 - res_pjsip_session: Initial INVITE with
audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
* ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
58(Bearer capability not available)
(Reported by Jared Hull)
* ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
if remote leg has T.38 disabled
(Reported by Torrey Searle)
* ASTERISK-26686 - res_pjsip: Lock inversion in transport management
(Reported by Ross Beer)
* ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
(Reported by Joshua Elson)
Improvements made in this release:
-----------------------------------
* ASTERISK-28006 - PJSIP: Missing
"party=calling"/"party=called" in Remote-Party-ID
(Reported by Eric Dantie)
* ASTERISK-27995 - pjproject_bundled: Find shared libraries in
root --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27993 - pjsip_wizard example gives wrong info about
unsupported SRV records
(Reported by Jonathan Harris)
* ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
backspace or end of line are merged with regular text and it
causes some UA to break
(Reported by Emmanuel BUU)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.23.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.22.0.
The release of Asterisk 13.22.0 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-27818 - Username bruteforce is possible when using
ACL with PJSIP
(Reported by John)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown
(Reported by Kevin Harwell)
* ASTERISK-27870 - app_confbridge: Conference bridge and
announcer channels are not removed if conference is ended as
soon as it starts
(Reported by Robert Mordec)
* ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
and submit_unscheduled_batch
(Reported by Denis Lebedev)
* ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
module pbx_dundi.so with dundi peers
(Reported by Kirsty Tyerman)
* ASTERISK-27943 - AMI: Action SendText needs to use the
correct thread.
(Reported by Richard Mudgett)
* ASTERISK-27942 - res_pjsip_messaging doesn't accept
application/* content-types.
(Reported by George Joseph)
* ASTERISK-27936 - res_pjsip_session doesn't update media when
a 200 comes in with a different port than a 183
(Reported by George Joseph)
* ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)
* ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
POSIX compatible.
(Reported by Alexander Traud)
* ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
parameter passed and aliased.
(Reported by Alexander Traud)
* ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27705 - chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)
* ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27888 - SQL fetch error on query which return 0 columns
(Reported by Alexei Gradinari)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses
(Reported by George Joseph)
* ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
before terminating nul.
(Reported by Alexander Traud)
* ASTERISK-27094 - res_fax: Deadlock when using Local channels
and fax gateway
(Reported by David Brillert)
* ASTERISK-25261 - Manager events for MeetMe have incorrectly
documented key name 'Usernum' - should be 'User'
(Reported by Francois Blackburn)
* ASTERISK-27878 - [patch] tcptls.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27872 - res_pjsip: Modified qualify_frequency
doesn't effect until pjsip reload
(Reported by Alexei Gradinari)
* ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh.
(Reported by Alexander Traud)
* ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
configured with enable-ssl3-method no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
marker bit error
(Reported by Torrey Searle)
* ASTERISK-27863 - config/ast_destroy_realtime_fields:
successful DELETE is treated as failed
(Reported by Alexei Gradinari)
* ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27853 - Incorrect error reported when
leaving/retrieving a ODBC voicemail
(Reported by Nic Colledge)
* ASTERISK-27726 - chan_mobile: presents incorrect inbound Caller-ID names
(Reported by Brian)
* ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
Unregister the module for headers.
(Reported by Alexander Traud)
* ASTERISK-27860 - [patch] res_pjsip: Register
pjsip_transport_management not externally but internally.
(Reported by Alexander Traud)
* ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
with recent MariaDB version.
(Reported by Nic Colledge)
* ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
timeout as being minutes.
(Reported by Corey Farrell)
* ASTERISK-27824 - Fix issues exposed by GCC 8
(Reported by George Joseph)
* ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility.
(Reported by Alexander Traud)
* ASTERISK-27841 - digest over for manager (ami) over http
fails on too long uris
(Reported by Jaco Kroon)
* ASTERISK-26570 - Macro allows an infinite loop of dialplan
inclusion resulting in a crash
(Reported by Tzafrir Cohen)
* ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on"
(Reported by shaurya jain)
* ASTERISK-26806 - pjsip_options: rework to make more efficient
(Reported by Kevin Harwell)
* ASTERISK-27814 - translate: interpolated frames are not passed through
(Reported by Kevin Harwell)
* ASTERISK-27812 - When the ooh323 debug is on there is no
ringing signal to incoming calls via H323 trunk.
(Reported by Dimos)
* ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
debug is enabled only on the module
(Reported by Marco Giordani)
* ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
designator extension.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27752 - Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)
* ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27877 - app_confbridge: Add talking indicator for
ConfBridgeList AMI response
(Reported by William McCall)
* ASTERISK-27873 - documentation: Error on wiki description of
Asterisk 13 "MeetmeMute" event
(Reported by Alessandro Polidori)
* ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)
* ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27796 - res_hep: Allow create_address to resolve a
provided hostname
(Reported by Sebastian Gutierrez)
* ASTERISK-27820 - [patch] Add DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27793 - cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.22.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 15, 13 and 14, and Certified Asterisk 13.18
and 13.21. The available releases are released as versions 15.4.1,
13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2018-007: Infinite loop when reading iostreams
When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
sends a specially crafted message then Asterisk gets caught in an infinite
loop while trying to read the data stream. Thus rendering the system as
unusable.
* AST-2018-008: PJSIP endpoint presence disclosure when using ACL
When endpoint specific ACL rules block a SIP request they respond with a 403
forbidden. However, if an endpoint is not identified then a 401 unauthorized
response is sent. This vulnerability just discloses which requests hit a
defined endpoint. The ACL rules cannot be bypassed to gain access to the
disclosed endpoints.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.21.0.
The release of Asterisk 13.21.0 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:
-----------------------------------
* ASTERISK-27704 - Add cache_pools debug option to pjproject.conf
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for
compiler clang conditionally.
(Reported by Alexander Traud)
* ASTERISK-27774 - res_musiconhold: Music on hold restarts
after every announcement
(Reported by lvl)
* ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition
(Reported by Evandro César Arruda)
* ASTERISK-27614 - res_pjsip_session: SDP origin does not use
resolved address
(Reported by John M.)
* ASTERISK-27740 - chan_sip: New Channel creation from new SIP
dialog with Replaces failed to be properly tracked and destroyed
(Reported by Shannon Price)
* ASTERISK-27706 - PJSIP: Deadlock shutting down subscription
TCP connection and sending subscription message.
(Reported by Ross Beer)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27761 - [patch] BuildSystem: With external editline,
do not require libs for internal editline.
(Reported by Alexander Traud)
* ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put
on hold and clear talking status
(Reported by Kevin Harwell)
* ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport Disconnect
(Reported by Ross Beer)
* ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on
a channel are equal
(Reported by George Joseph)
* ASTERISK-27745 - [patch] BuildSystem: Remove unused
dependency on libltdl.
(Reported by Alexander Traud)
* ASTERISK-12841 - [patch] Make format_ogg_vorbis work on OpenBSD
(Reported by Michiel van Baak)
* ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux
Sound Architecture (ALSA) in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27741 - res_pjsip_rfc3326.c
rfc3326_use_reason_header doesn't account for more than one
'Reason' header
(Reported by Ross Beer)
* ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on
openSUSE and Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
(Reported by Alexander Traud)
* ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25
(Reported by Curt Sampson)
* ASTERISK-27641 - BuildSystem: Enable Better Backtraces in FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-25586 - uuid_generate_random detection failure
(Reported by John Nemeth)
* ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to
colon character when not found.
(Reported by Alexander Traud)
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0 MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-27674 - chan_sip: RTP framing issues on outgoing calls
(Reported by Jean Aunis - Prescom)
* ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers
break many endpoints
(Reported by Ross Beer)
* ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly
but explicitly on external libraries.
(Reported by Alexander Traud)
* ASTERISK-27719 - [patch] res_http_post: Enable GMime in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27716 - [patch] BuildSystem: Enable autotools in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27714 - [patch] chan_unistim: NetBSD has an
incompatible struct in_pktinfo.
(Reported by Alexander Traud)
* ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t
explicitly to its proposed type.
(Reported by Alexander Traud)
* ASTERISK-27712 - [patch] BuildSystem: Detect whether
uselocale(.) is available.
(Reported by Alexander Traud)
* ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of
pthread_* on NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27710 - [patch] BuildSystem: Install init scripts on
openSUSE Tumbleweed.
(Reported by Alexander Traud)
* ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison
in ./configure.
(Reported by Alexander Traud)
* ASTERISK-27610 - app_amd.so returning TOOLONG before reaching
the timeout
(Reported by Michael Cargile)
* ASTERISK-26688 - Documentation: voicemail.conf.sample shows
512 limit for emailbody field, however this is only true if
compiled with LOW_MEMORY option
(Reported by Fran Vicente)
* ASTERISK-27568 - PJSIP: Crash during SIP attended transfer.
(Reported by Bryan Walters)
* ASTERISK-27686 - [patch] install_prereq: Update FreeBSD libraries.
(Reported by Alexander Traud)
* ASTERISK-24488 - Wrong remote identity and target in dialog
package XML in NOTIFY
(Reported by Alejandro Padilla)
* ASTERISK-27646 - ICE fails with no candidate nominated
(Reported by Thomas Guebels)
* ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS)
if existing peer from same IP.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip channels
(Reported by Nathan Bruning)
* ASTERISK-26540 - cdr_radius: use radcli instead of
freeradius-client
(Reported by Tzafrir Cohen)
* ASTERISK-27770 - [patch] install_prereq: Add Slackware (somehow).
(Reported by Alexander Traud)
* ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux.
(Reported by Alexander Traud)
* ASTERISK-27738 - [patch] install_prereq: Add Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27736 - [patch] install_prereq: Add SUSE.
(Reported by Alexander Traud)
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by Alex)
* ASTERISK-27728 - [patch] BuildSystem: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version 2.7.2
(Reported by Richard Mudgett)
* ASTERISK-27729 - [patch] install_prereq: Add NetBSD.
(Reported by Alexander Traud)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.21.0
-----
The release of Asterisk 13.20.0 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-27583 - Segmentation fault occurs in asterisk with
an invalid SDP fmtp attribute
(Reported by Sandro Gauci)
* ASTERISK-27582 - Segmentation fault occurs in Asterisk with
an invalid SDP media format description
(Reported by Sandro Gauci)
* ASTERISK-27618 - Crash occurs when sending a repeated number
of INVITE messages over TCP or TLS transport
(Reported by Sandro Gauci)
* ASTERISK-27640 - SUBSCRIBE message with a large Accept value
causes stack corruption
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0
MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt"
needs update or removal.
(Reported by Rusty Newton)
* ASTERISK-27689 - [patch] rtp_engine: Load format name / mime
type in uppercase again.
(Reported by Alexander Traud)
* ASTERISK-27679 - res_pjsip: Endpoint destruction does not
free DTLS configuration
(Reported by Mak Dee)
* ASTERISK-27684 - [patch] install_prereq: Update OpenBSD libraries.
(Reported by Alexander Traud)
* ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27680 - [patch] res_calendar: Specialized calendars
depend on symbols of general calendar.
(Reported by Alexander Traud)
* ASTERISK-27677 - [patch] BuildSystem: Enable system provided
libedit on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323 leftovers.
(Reported by Alexander Traud)
* ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with previous paths.
(Reported by Alexander Traud)
* ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash
is not installed.
(Reported by Alexander Traud)
* ASTERISK-27666 - chan_sip: Crash processing CANCEL request
(Reported by Leandro Dardini)
* ASTERISK-27584 - Internal pjproject build doesn't disable bcg729
(Reported by Stuart Henderson)
* ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC 2.0.
(Reported by Alexander Traud)
* ASTERISK-27642 - [patch] backtrace: Avoid
-Wlogical-not-parentheses.
(Reported by Alexander Traud)
* ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu libraries.
(Reported by Alexander Traud)
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
* ASTERISK-27426 - chan_console: cannot read and write at the
same time with alsa backend
(Reported by Tzafrir Cohen)
* ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI event
(Reported by sungtae kim)
* ASTERISK-27652 - Null pointer Crash in PJSIP MWI
(Reported by Joshua Elson)
* ASTERISK-27612 - Subscriptions Persist After Expiration and
TCP/TLS Disconnect
(Reported by Ross Beer)
* ASTERISK-27571 - res_pjsip: If SIP response is received
during shutdown a crash may occur
(Reported by Joshua Colp)
* ASTERISK-27637 - [patch] BuildSystem: Enable autotools in FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27635 - [patch] app_voicemail: Avoid always true
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-27599 - [patch] install_prereq: Update
RHEL/CentOS/Fedora libraries.
(Reported by Alexander Traud)
* ASTERISK-26563 - core: macOS devmode build fails: variable
'freeswap' set but not used
(Reported by David M. Lee)
* ASTERISK-27630 - [patch] editline: Avoid shifting a negative signed value.
(Reported by Alexander Traud)
* ASTERISK-16172 - Problems with siren14 codec; problems with
siren7 sound files.
(Reported by Steve Murphy)
* ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
(Reported by Stéphan Kochen)
* ASTERISK-27603 - [patch] install_prereq: Download latest Jansson.
(Reported by Alexander Traud)
* ASTERISK-27607 - [patch] res_config_mysql: Avoid the header mysql_version.h.
(Reported by Alexander Traud)
* ASTERISK-24598 - When running
./contrib/scripts/install_prereq install-unpackaged pjproject is
installed in wrong place
(Reported by PowerPBX)
* ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
(Reported by Alexander Traud)
* ASTERISK-27600 - [patch] BuildSystem: Allow make clean all again.
(Reported by Alexander Traud)
* ASTERISK-27598 - [patch] install_prereq: Support package manager DNF.
(Reported by Alexander Traud)
* ASTERISK-26596 - Placing call on hold temporarily locks up set
(Reported by Igor Goncharovsky)
* ASTERISK-27596 - [patch] BuildSystem: Use the detected name
for MD5 everywhere.
(Reported by Alexander Traud)
* ASTERISK-27594 - [patch] BuildSystem: Invoke install not in
GNU but POSIX style.
(Reported by Alexander Traud)
* ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
(Reported by Alexander Traud)
* ASTERISK-27592 - [patch] BuildSystem: Detect external library
Lua in version 5.3.
(Reported by Alexander Traud)
* ASTERISK-26832 - res_pjsip: Segfault when calling
pjsip_hdr_print_on in sip_msg.c:581
(Reported by Ross Beer)
* ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id -u instead.
(Reported by Alexander Traud)
* ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES
compiler flag
(Reported by Jean Aunis - Prescom)
* ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting
an int to unsigned char.
(Reported by Alexander Traud)
* ASTERISK-27560 - [patch] clang 5 does not know
-Wno-format-truncation
(Reported by Alexander Traud)
* ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)
* ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an
int to unsigned short.
(Reported by Alexander Traud)
* ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only
matches against header if match by ip fails
(Reported by George Joseph)
* ASTERISK-27549 - [patch] translate: Avoid absolute value on
unsigned substraction.
(Reported by Alexander Traud)
* ASTERISK-27553 - [patch] res_curl: Avoid error message on unload.
(Reported by Alexander Traud)
* ASTERISK-27557 - [patch] clang 5.0: implicit conversion to
char changes value to negative.
(Reported by Alexander Traud)
* ASTERISK-27559 - [patch] editline: Avoid comparison between
pointer and zero character constant.
(Reported by Alexander Traud)
* ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative signed value.
(Reported by Alexander Traud)
* ASTERISK-25329 - Asterisk configure fails on 'cannot find
ptlib-config', despite ptlib-config existing
(Reported by Rusty Newton)
* ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to
positive values as intended.
(Reported by Alexander Traud)
* ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header guard.
(Reported by Alexander Traud)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled.
(Reported by Tzafrir Cohen)
* ASTERISK-27498 - ICE candidate parser - ICE foundation
parsing too short
(Reported by Michele Pr)
* ASTERISK-27366 - Asterisk Turkish Language Set Problem
(Reported by Halil Ä°brahim YILDIZ)
* ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
(Reported by Shane Mitchell)
* ASTERISK-27531 - Compiler optimizations can break module load sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-24198 - Typo's
(Reported by Walter Doekes)
* ASTERISK-27229 - bridge: Old channel video source not set to
NULL after unref
(Reported by Richard Kenner)
Improvements made in this release:
-----------------------------------
* ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin
and channel snapshot headers to ConfbridgeList AMI events
(Reported by Richard Mudgett)
* ASTERISK-27647 - app_confbridge/bridge_softmix: When channel
muted report talking stopped if was talking.
(Reported by Richard Mudgett)
* ASTERISK-27084 - Reduce verbosity while loading PBX extensions.
(Reported by Ludovic Gasc (Eyepea))
* ASTERISK-24372 - [patch] Add config option to play a prompt
to the "winner" in app_followme
(Reported by Graham Mainwaring)
* ASTERISK-27461 - 3PCC patch for AMI "SIPnotify"
(Reported by Yasuhiko Kamata)
* ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate
from chan_sip to chan_pjsip realtime
(Reported by Torrey Searle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.20.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.19.2, 14.7.6,
15.2.2 and 13.18-cert3.
The following security vulnerabilities were resolved in these versions:
* AST-2018-001: Crash when receiving unnegotiated dynamic payload
The RTP support in Asterisk maintains its own registry of dynamic codecs and
desired payload numbers. While an SDP negotiation may result in a codec using
a different payload number these desired ones are still stored internally.
When an RTP packet was received this registry would be consulted if the
payload number was not found in the negotiated SDP. This registry was
incorrectly consulted for all packets, even those which are dynamic. If the
payload number resulted in a codec of a different type than the RTP stream
(for example the payload number resulted in a video codec but the stream
carried audio) a crash could occur if no stream of that type had been
negotiated. This was due to the code incorrectly assuming that a stream of the
type would always exist.
* AST-2018-002: Crash when given an invalid SDP media format description
By crafting an SDP message with an invalid media format description Asterisk
crashes when using the pjsip channel driver because pjproject's sdp parsing
algorithm fails to catch the invalid media format description.
* AST-2018-003: Crash with an invalid SDP fmtp attribute
By crafting an SDP message body with an invalid fmtp attribute Asterisk
crashes when using the pjsip channel driver because pjproject's fmtp retrieval
function fails to check if fmtp value is empty (set empty if previously parsed
as invalid).
* AST-2018-004: Crash when receiving SUBSCRIBE request
When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
accepted formats present in the Accept headers of the request. This code did
not limit the number of headers it processed despite having a fixed limit of
32. If more than 32 Accept headers were present the code would write outside
of its memory and cause a crash.
* AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
A crash occurs when a number of authenticated INVITE messages are sent over
TCP or TLS and then the connection is suddenly closed. This issue leads to a
segmentation fault.
* AST-2018-006: WebSocket frames with 0 sized payload causes DoS
When reading a websocket, the length was not being checked. If a payload of
length 0 was read, it would result in a busy loop that waited for the
underlying connection to close.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-----
The release of Asterisk 13.19.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.1
Revision 1.65 / (download) - annotate - [select for diffs], Thu Nov 5 09:07:38 2020 UTC (2 years, 7 months ago) by ryoon
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asterisk*: Fix install on SunOS.
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update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
-----------------------------------
* ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
incoming INVITE Request-URI.
(Reported by Richard Mudgett)
* ASTERISK-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* ASTERISK-27382 - crash after an invalid rtcp packet from GT48
FXS gateway
(Reported by Tzafrir Cohen)
* ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
RTCP packet will write past where it should
(Reported by Vitezslav Novy)
* ASTERISK-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if
flooded with unauthenticated requests
(Reported by George Joseph)
* ASTERISK-27454 - res_http_post: Don't require
GMIME_MAJOR_VERSION
(Reported by Joshua Colp)
* ASTERISK-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* ASTERISK-27411 - pjsip: TCP connections may not be destroyed
(Reported by Joshua Colp)
* ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
responses.
(Reported by Corey Farrell)
* ASTERISK-27337 - chan_sip: Security vulnerability with client
code header (revisited)
(Reported by Richard Mudgett)
* ASTERISK-27319 - (Security) Function in PJSIP 2.7
miscalculates the length of an unsigned long variable in 64bit
machines
(Reported by Kim youngsung)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
* ASTERISK-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-27278 - [patch] chan_sip: Provide access to read the
full SIP Request-URI from INVITE
(Reported by David J. Pryke)
* ASTERISK-27255 - alembic: Add support for Microsoft SQL
server
(Reported by Florian Floimair)
* ASTERISK-27253 - [patch] libsrtp-2.1.x support
(Reported by Alexander Traud)
* ASTERISK-27220 - Enable CHANNEL function to get from and to
tag from SIP Headers
(Reported by Andre Nazario)
* ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)
* ASTERISK-27173 - Support for GMIME 3.0
(Reported by Tzafrir Cohen)
* ASTERISK-27092 - [patch] app_queue: Add Priority to AMI
QueueStatus
(Reported by Niklas Larsson)
* ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
chan_pjsip
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
caller-id when it shouldn't be.
(Reported by dtryba)
* ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
user=phone parameters to URIs
(Reported by dtryba)
* ASTERISK-27270 - cdr_mysql: various crashes at second module
reload if cdr_mysql.conf is configured
(Reported by Tzafrir Cohen)
* ASTERISK-27301 - [patch] app_queue: Music On Hold for
real-time queues is not reset to default
(Reported by Nathan Bruning)
* ASTERISK-25266 - Application Originate returns SUCCESS to
ORIGINATE_STATUS upon failure to originate
(Reported by Allen Ford)
* ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
unavailable endpoints
(Reported by Richard Mudgett)
* ASTERISK-27305 - res_ari: Memory leaks in ARI when using
Content-Type: application/json
(Reported by David Hajek)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
IPv4 client via TCP/TLS
(Reported by Alexander Traud)
* ASTERISK-27317 - vector: multiple evaluation of elem in
AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
* ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
ast_strings_match
(Reported by Corey Farrell)
* ASTERISK-27296 - [patch] False positive busy checks when
icalendar's recurrence-id mechanism is involved
(Reported by Benoît Dereck-Tricot)
* ASTERISK-27284 - Status of RFC 3323 and PJSIP
(Reported by dtryba)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engström)
* ASTERISK-27295 - Contact is improperly translated after
d178f497
(Reported by Sean Bright)
* ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
(SSRC Changes)
(Reported by Ross Beer)
* ASTERISK-27289 - A codeblock that maintains a bug,but maybe
the codeblock will never run
(Reported by Huangyx)
* ASTERISK-27283 - Realtime config fail with PostgreSQL version
before 9.1
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-27257 - bridge_native_rtp: half-way direct media
when using early bridging
(Reported by Jean Aunis - Prescom)
* ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
Possible PJSIP Vulnerability
(Reported by Ross Beer)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-16898 - SRTP unprotect: authentication failure when
RTP sequence number switches from 65535 -> 0
(Reported by Marcello Ceschia)
* ASTERISK-27274 - RTCP needs better packet validation to
resist port scans.
(Reported by Richard Mudgett)
* ASTERISK-27252 - RTP: One way audio with direct media and
strictrtp=yes.
(Reported by Richard Mudgett)
* ASTERISK-25524 - module reload res_calendar.so does not
reload everything in calendar.conf
(Reported by Jesper)
* ASTERISK-24588 - res_calendar does not process CalDAV from
Owncloud [fix included]
(Reported by Stefan Gofferje)
* ASTERISK-25523 - res_calendar: Warning about invalid channel
value (for notification) occurs even when event has no
notification configured.
(Reported by Jesper)
* ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
wireshark disagree
(Reported by Tzafrir Cohen)
* ASTERISK-27248 - [patch]external_media_address and
external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
* ASTERISK-27165 - CDR: CDR(start,u) function won't work in
cdr_custom config
(Reported by Jacek Konieczny)
* ASTERISK-27217 - chan_sip: Asterisk crashing when
subscription doesn't get set
(Reported by Bryan Walters)
* ASTERISK-24066 - res_smdi: convert to astobj2
(Reported by Corey Farrell)
* ASTERISK-17540 - SDP origin attribute modified when issuing
re-INVITE because of directmedia=yes
(Reported by saghul)
* ASTERISK-27254 - alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
* ASTERISK-27232 - When in queue on g722 with interruptions,
music on hold can get stuck and no longer play
(Reported by Jens T.)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
* ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
* ASTERISK-27177 - ooh323c: misleading indentation in
addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
* ASTERISK-27241 - libc segfault upon entry into app_directory
(Reported by David Moore)
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-26994 - Confbridge: CBAnn channels intermittently
become stuck when caller hangs up before recording name
(Reported by James Terhune)
* ASTERISK-20858 - app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
* ASTERISK-16777 - several filename bugs in Record()
application
(Reported by klaus3000)
* ASTERISK-27168 - alembic: PJSIP scripts are missing column
dtls_fingerprint in ps_endpoints table
(Reported by Florian Floimair)
* ASTERISK-23608 - ControlPlayback fails to play files with
names containing certain non-alpha characters
(Reported by Jonathan White)
* ASTERISK-19103 - When using realtime queues, function
QUEUE_MEMBER_LIST() will return an error if no other
app/function has loaded the queues first. This problem does not
exist if queues.conf is used.
(Reported by Jim Van Meggelen)
* ASTERISK-21241 - When using voicemail as announce only
(maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
* ASTERISK-27204 - [patch] app_queue: Wrong queue stat
calculation
(Reported by sungtae kim)
* ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
is used
(Reported by Torrey Searle)
* ASTERISK-27207 - XMPP OAuth not working due to inverted
logic
(Reported by Michael Kuron)
* ASTERISK-27174 - res_calendar_icalendar: Recurring events not
being loaded from Google calendar using ical
(Reported by Mark Thompson)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Seán C. McCord)
* ASTERISK-27147 - Either asterisk or pjproject isn't re-using
tcp connections (again)
(Reported by George Joseph)
* ASTERISK-27193 - IPv6 receive address in message doesn't
include brackets
(Reported by Scott Griepentrog)
* ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
are not available when native bridge is used
(Reported by Torrey Searle)
* ASTERISK-27110 - RTP session is not fully destroyed on
channel hangup
(Reported by Matt Jordan)
* ASTERISK-26745 - Asymmetric codecs when
asymmetric_rtp_codec=no
(Reported by Jesse Ross)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
around the status element in XML
(Reported by Abraham Liebsch)
* ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
devmode enabled.
(Reported by Corey Farrell)
* ASTERISK-27130 - Applications ARI: Unsubscribe action for
deviceStates does not remove old subscriptions properly
(Reported by Sergej Kasumovic)
* ASTERISK-25810 - say.c calls for sounds in the subdir
"digits" that don't exist (in Core). SayUnixTime or other Say...
apps will fail out when they call these sounds.
(Reported by Nicolas Riendeau)
* ASTERISK-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-27124 - app_playback.c:say_date_generic use
timezonename parameter
(Reported by Holger Hans Peter Freyther)
* ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
RTCP-MUX in use
(Reported by Joshua Colp)
* ASTERISK-27128 - [patch]res_stasis_snoop: When recording a
snoop channel (using ARI) where no media is being received, no
recording happens when theres no media
(Reported by Dan Jenkins)
* ASTERISK-27123 - confbridge: Name recordings are left on
filesystem
(Reported by Sergej Kasumovic)
* ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep
adding up
(Reported by Sergej Kasumovic)
* ASTERISK-26807 - sounds: New 3-D Binaural audio features
require new sound prompts
(Reported by Rusty Newton)
* ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts
differ in content from the English files
(Reported by Benoit Duverger)
* ASTERISK-26274 - Resolve open sounds issues and then create a
new sounds release (1.5.1? or 1.6?)
(Reported by Rusty Newton)
* ASTERISK-27127 - configs: Erroneous load directive in sample
configuration results in "Error loading module
'res_pjsip_multihomed.so'"
(Reported by HZMI8gkCvPpom0tM)
* ASTERISK-27105 - [patch]core: when setting 'maxfiles' in
asterisk.conf, a message is printed, even in rasterisk -x
(Reported by Tzafrir Cohen)
* ASTERISK-27036 - res_pjsip: Asterisk crashes when an
extension tries to use PJSIP trunk with from_user containing
'@'
(Reported by Maxim Vasilev)
* ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session
in use
(Reported by Jatin Jain)
* ASTERISK-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27108 - Crash using 'data get' CLI command
(Reported by Sean Bright)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
only really different domain with TLS.
(Reported by Alexander Traud)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to
clear flag in an error branch.
(Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport
(Reported by Richard Mudgett)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue
(Reported by Marek Cervenka)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()
(Reported by Ross Beer)
* ASTERISK-27074 - core_local: local channel data not being
properly unref'ed and unlocked
(Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed
attended transfer
(Reported by Kevin Harwell)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
RTCP component
(Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels
are added to bridge
(Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
during transfer
(Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow
network
(Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when
channel from endpoint is in early media
(Reported by Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to
'Unreachable' endpoints
(Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and
snooping
(Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime
(Reported by Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal
Server Error
(Reported by Tim Morgan)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
in wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-23951 - Asterisk attempts and fails to build
format_mp3 even if mp3lib was not downloaded
(Reported by Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
describe BEEP argument
(Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
variable" command without args
(Reported by Antoine Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
* ASTERISK-25101 - DTLS configuration can not be specified in
the general section - documentation
(Reported by Ben Langfeld)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
fmtp optional parameters have a space
(Reported by John Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is
parked
(Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called
after AMI "Redirect" action for queues with wrapuptime
(Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new
calls after doing a transfer if wrapuptime = greater than 0 and
using Local channel
(Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
agents not to receive queue calls after transfer queue call
(Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
not play user name recording while leaving
(Reported by Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier Riveros)
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
with LibreSSL
(Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on
FreeBSD due to missing crypt.h file
(Reported by Guido Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec
translations when using audiohooks
(Reported by Michael Walton)
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by Alex)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
Revision 1.32 / (download) - annotate - [select for diffs], Mon Jan 1 21:18:17 2018 UTC (5 years, 5 months ago) by adam
Branch: MAIN
Changes since 1.31: +2 -2
lines
Diff to previous 1.31 (colored)
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Revision 1.31 / (download) - annotate - [select for diffs], Thu Nov 30 16:45:16 2017 UTC (5 years, 6 months ago) by adam
Branch: MAIN
CVS Tags: pkgsrc-2017Q4-base,
pkgsrc-2017Q4
Changes since 1.30: +2 -2
lines
Diff to previous 1.30 (colored)
Revbump after textproc/icu update
Revision 1.30 / (download) - annotate - [select for diffs], Mon Sep 18 09:53:12 2017 UTC (5 years, 8 months ago) by maya
Branch: MAIN
CVS Tags: pkgsrc-2017Q3-base,
pkgsrc-2017Q3
Changes since 1.29: +2 -2
lines
Diff to previous 1.29 (colored)
revbump for requiring ICU 59.x
Revision 1.29 / (download) - annotate - [select for diffs], Thu Aug 24 20:03:07 2017 UTC (5 years, 9 months ago) by adam
Branch: MAIN
Changes since 1.28: +2 -1
lines
Diff to previous 1.28 (colored)
Revbump for boost update
Revision 1.28 / (download) - annotate - [select for diffs], Sun Jun 4 07:51:27 2017 UTC (6 years ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2017Q2-base,
pkgsrc-2017Q2
Changes since 1.27: +2 -2
lines
Diff to previous 1.27 (colored)
Update to Asterisk 13.16.0: this is mostly a bugfix release.
The Asterisk Development Team would like to announce the release
of Asterisk 13.16.0.
The release of Asterisk 13.16.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engström)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier Riveros)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-26998 - res_pjsip_session: INVITE retransmissions
could still setup the same call again.
(Reported by Richard Mudgett)
* ASTERISK-26143 - res_rtp_asterisk: One way audio when
transcoding
(Reported by Henning Holtschneider)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-26983 - Crash in Manager Reload when TLS Config
Changes
(Reported by Joshua Elson)
* ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with
wrong eventtime
(Reported by Etienne Lessard)
* ASTERISK-26173 - func_cdr: CDR function does not permit empty
values to be assigned
(Reported by gkloepfer)
* ASTERISK-25506 - [patch]CONFBRIDGE failure after an
app_confbrige.so module reload results in segfault or
error/warning messages.
(Reported by Frederic LE FOLL)
* ASTERISK-24529 - Using AMI Action Bridge to on an already
bridged channel causes the incorrect return priority to be used
(Reported by Corey Farrell)
* ASTERISK-26860 - Upon RTCP reception, netsock2.c:210
ast_sockaddr_split_hostport: Port missing in (null)
(Reported by Evers Lab)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
* ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a
res_pjsip session to be leaked.
(Reported by Richard Mudgett)
* ASTERISK-25823 - SIGSEGV, Segmentation fault. -
../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Krüger)
* ASTERISK-26951 - chan_sip: ACK with SDP does not update a
direct media bridge
(Reported by Jean Aunis - Prescom)
* ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
fails for non-SSE2 instrunction Linux
(Reported by abelbeck)
* ASTERISK-26926 - func_speex: Crash caused by frame with no
datalen
(Reported by Richard Kenner)
* ASTERISK-26929 - pjsip: Add database tables for RLS
(Reported by Joshua Colp)
* ASTERISK-26953 - Asterisk crash if hep.conf have some missing
parameters
(Reported by Joel Vandal)
* ASTERISK-26890 - STUN server with non-default-route transport
causes INVITE delay
(Reported by George Joseph)
* ASTERISK-26692 - res_rtp_asterisk: Crash in
dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
(Reported by scgm11)
* ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP
address string
(Reported by Niklas Larsson)
* ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when
receiving packet
(Reported by Adagio)
* ASTERISK-26613 - format_wav: wav16 format read file only by
320 - half of frame
(Reported by Vitaly K)
* ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in
MixMonitor
(Reported by Ivan Myalkin)
* ASTERISK-21856 - STUN never works when asterisk started
without internet access
(Reported by Jeremy Kister)
* ASTERISK-20984 - Audible clicks when playing sox encoded au
file with STREAM FILE AGI command
(Reported by Roman S.)
* ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
same IP as explicit transport
(Reported by Richard Begg)
* ASTERISK-26903 - Listening TCP/TLS sockets stop when
temporarily out of open files
(Reported by Walter Doekes)
* ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in
ast_str_case_hash
(Reported by Badalian Vyacheslav)
* ASTERISK-26928 - pjsip: Add database tables for PUBLISH
support
(Reported by Joshua Colp)
* ASTERISK-26927 - pjproject_bundled: Crash on
pj_ssl_get_info() while ioqueue_on_read_complete().
(Reported by Alexander Traud)
* ASTERISK-26905 - pjproject_bundled: Merge 3 upstream
deadlock patches into bundled
(Reported by Ross Beer)
* ASTERISK-26897 - chan_sip: Security vulnerability with client
code header
(Reported by Alex VillacÃs Lasso)
* ASTERISK-25974 - Unused realtime MOH classes not purged on
'moh reload'
(Reported by Sébastien Couture)
* ASTERISK-26916 - res_pjsip: Excessive refcount reached on
transport ao2 object
(Reported by Ross Beer)
* ASTERISK-21721 - SIP Failed to parse multiple Supported:
headers
(Reported by Olle Johansson)
* ASTERISK-26915 - chan_sip: Session Timers required but
refused wrongly.
(Reported by Alexander Traud)
* ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not
authenticated even after receiving a 407 error code
(Reported by Yaacov Akiba Slama)
* ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn
with large app_args causes ABRT
(Reported by twisted)
* ASTERISK-26705 - libasteriskssl.so not found when asterisk is
installed for the 1st time
(Reported by George Joseph)
* ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ
when creating pubsub unsubscription on client
(Reported by
Marcello Ceschia)
* ASTERISK-25490 - [patch]SDP crypto tag is validated
incorrectly
(Reported by Joerg Sonnenberger)
* ASTERISK-24712 - xmpp: starttls problem causes connection
spew
(Reported by Matthias Urlichs)
* ASTERISK-26086 - res_musiconhold: format option is not
documented adequately
(Reported by Jens Bürger)
* ASTERISK-23996 - No core dumps because of res_musiconhold
chdir.
(Reported by Walter Doekes)
* ASTERISK-26814 - pjproject_bundled build fails to download
pjproject source when using cURL
(Reported by Gergely Dömsödi)
* ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for
unavailable clients
(Reported by Anthony Critelli)
* ASTERISK-21855 - Asterisk crashes when XMPP message is sent
(JabberSend) and no internet connection is available
(Reported by Jeremy Kister)
* ASTERISK-25622 - WARNING for "JABBER: socket read error"
should be more specific
(Reported by Sean Darcy)
* ASTERISK-26818 - cdr: Problem setting variables in h exten
(Reported by scgm11)
* ASTERISK-26875 - app_mixmonitor: Recording out of sync when
183 but no RTP
(Reported by Aaron An)
Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0
Thank you for your continued support of Asterisk!
Revision 1.27 / (download) - annotate - [select for diffs], Mon May 29 20:52:37 2017 UTC (6 years ago) by jnemeth
Branch: MAIN
Changes since 1.26: +2 -2
lines
Diff to previous 1.26 (colored)
Add fixes for AST-2017-002, AST-2017-003, and AST-2017-004. Note that the first two don't affect pkgsrc as we are using chan_sip not PJSIP. The last only affects users of SCCP, which is Cisco's proprietary protocol. ----- AST-2017-002 A remote crash can be triggered by sending a SIP packet to Asterisk with a specially crafted CSeq header and a Via header with no branch parameter. The issue is that the PJSIP RFC 2543 transaction key generation algorithm does not allocate a large enough buffer. By overrunning the buffer, the memory allocation table becomes corrupted, leading to an eventual crash. This issue is in PJSIP, and so the issue can be fixed without performing an upgrade of Asterisk at all. However, we are releasing a new version of Asterisk with the bundled PJProject updated to include the fix. If you are running Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-003 The multi-part body parser in PJSIP contains a logical error that can make certain multi-part body parts attempt to read memory from outside the allowed boundaries. A specially-crafted packet can trigger these invalid reads and potentially induce a crash. The issue is within the PJSIP project and not in Asterisk. Therefore, the problem can be fixed without upgrading Asterisk. However, we will be releasing a new version of Asterisk where the bundled version of PJSIP has been updated to have the bug patched. If you are using Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-004 A remote memory exhaustion can be triggered by sending an SCCP packet to Asterisk system with chan_skinny enabled that is larger than the length of the SCCP header but smaller than the packet length specified in the header. The loop that reads the rest of the packet doesn't detect that the call to read() returned end-of-file before the expected number of bytes and continues infinitely. The partial data message logging in that tight loop causes Asterisk to exhaust all available memory.
Revision 1.26 / (download) - annotate - [select for diffs], Sat May 13 22:39:13 2017 UTC (6 years ago) by jnemeth
Branch: MAIN
Changes since 1.25: +7 -3
lines
Diff to previous 1.25 (colored)
Update to Asterisk 13.15.0. This is mostly a bug fix release with a few
minor enhancements. 13.14.1 was released to fix AST-2017-001.
----- 13.15.0
The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
The release of Asterisk 13.15.0 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:*
-----------------------------------
- [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)
*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacíó Lasso)
- [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/ASTERISK-26869>]
- res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
extension
(Reported by Torrey Searle)
- [ASTERISK-26668 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/ASTERISK-26872>]
- Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)
- [ASTERISK-26717 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>]
- pjsip/cli_commands: pjsip show channelstats shows wrong codec
(Reported by Kevin Harwell)
- [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>]
- res_pjsip: Crash when using IPv6 and Transport ws,wss
(Reported by Michael Balen)
- [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>]
- app_voicemail: Cannot set fromstring on a per-mailbox basis
(Reported by Mark Scholten)
- [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>]
- Saynumber is trying to get "and" from "digits/" subfolder
(Reported by Jonathan Harris)
- [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>]
- Long lines in call files cause spurious syntax error
(Reported by Dave Olszewski)
- [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>]
- res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
(Reported by Jøògen H)
- [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>]
- res_config_pgsql: should match the behavior of other drivers so that
queue_log can disable adaptive logging
(Reported by Dmitry Wagin)
- [ASTERISK-26825 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>]
- pjsip.conf.sample: user_agent: still refers to branch 12
(Reported by Tzafrir Cohen)
- [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>]
- PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not
exist
(Reported by Mark Michelson)
- [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>]
- res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by Jøògen H)
- [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>]
- res_pjsip_outbound_registration doesn't know about network change events
(Reported by George Joseph)
- [ASTERISK-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>]
- chan_sip : Asterisk restart seems to be required for changing encryption
option
(Reported by benasse)
- [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>]
- bridge: Passing the 'p' (play tone) flag to Bridge() application results
in garbled audio
(Reported by Sean Bright)
- [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>]
- res_pjsip: URI requirement for fields is not consistently documented and
error does not provide indication
(Reported by Peter Sokolov)
- [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>]
- [patch] Fix download_externals To Allow The Use Of curl Or wget
(Reported by Michael L. Young)
- [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>]
- Pattern matching with res_config_mysql extensions does not behave as
expected
(Reported by Charlie Smurthwaite)
- [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>]
- PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
- [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>]
- [patch] DUNDi weight parameter not processed correctly
(Reported by Peter Racz)
- [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>]
- [patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)
- [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>]
- res_pjsip: Using an auth object for inbound and outbound authentication
fails.
(Reported by Richard Mudgett)
- [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/ASTERISK-26738>]
- Frequent segfaults since activation of DNS SRV, in
pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and
pj_atomic_inc_and_get at pj/os_core_unix.c
(Reported by Michael Maier)
- [ASTERISK-25893 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>]
- Function vmauthenticate accesses uninitialized memory
(Reported by Filip Jenicek)
- [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>]
- [patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)
- [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>]
- [patch] Fix query with double backslash in string literals and stop log
warnings
(Reported by Humberto Figuera)
- [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>]
- res_config_sqlite3 uses incorrect query - unnecessary escape
(Reported by Stepan)
- [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>]
- SQlite3: Realtime queue loading fails after PRAGMA query result
(Reported by Scott Griepentrog)
- [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>]
- http: Crash on Reload Only in ast_tcptls_server_start
(Reported by Joshua Elson)
- [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>]
- Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)
- [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>]
- pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on
subscription refresh
(Reported by Zach R)
- [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>]
- res_pjsip_mwi: Asterisk does not terminate MWI subscription
(Reported by Carl Fortin)
- [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>]
- Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
- [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>]
- VoiceMailPlayMsg not playing messages via realtime
(Reported by Ryan Rittgarn)
- [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>]
- [patch] 'Silence' is truncated in Record()
(Reported by var)
- [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>]
- chan_pjsip: Error when calling PJSIP client with domain specified
(Reported by Norbert Varga)
- [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>]
- core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
- [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>]
- pbx: AMI Originate ignore "failed" extension on call failure
(Reported by Nasir Iqbal)
- [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>]
- configs/samples: The 'identify' entry is in the wrong section in
sorcery.conf.sample
(Reported by Torrey Searle)
- [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>]
- Crash in srv.c on startup with pjsip
(Reported by nappsoft)
- [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>]
- res_stasis_device_state: Duplicate subscriptions when multiple received
at same time
(Reported by Joshua Colp)
*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>]
- res_pjsip_session: Add support for overlap dialling
(Reported by Richard Begg)
- [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>]
- chan_sip: Add rtcp-mux support
(Reported by Sean Bright)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0
*Thank you for your continued support of Asterisk!*
----- 13.14.0
The Asterisk Development Team has announced the release of Asterisk 13.14.0.
The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
by nappsoft)
* ASTERISK-26704 - res_odbc.conf contains deprecated
configuration: 'pooling', 'shared_connections', 'limit', and
'idlecheck' options were replaced by 'max_connections'.
(Reported by Anthony Messina)
* ASTERISK-21094 - MixMonitorMute mutes through stream if already
slinear (e.g. Originate) (Reported by David Woolley)
* ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
hung up via ARI (Reported by Tom Pawelek)
* ASTERISK-26632 - core: Possibility of a frame "imbalance"
leading to stuck channels. (Reported by Mark Michelson)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
(Reported by Dmitriy)
* ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
request to endpoint (Reported by Ross Beer)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-26754 - build_tools: make_build_h does not handle \ in
user name (Reported by Kirill Katsnelson)
* ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
fwrite() returned error: Broken pipe" (Reported by Kirill
Katsnelson)
* ASTERISK-26755 - app_queue: Random queues disappear on "core
reload queue all" (Reported by Kirill Katsnelson)
* ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
after match in .conf has no effect (Reported by Michael Maier)
* ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
for SRV (Reported by Joshua Colp)
* ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
does not work. (Reported by Richard Mudgett)
* ASTERISK-26740 - voicemail API test: uses varlibdir instead of
datadir for a sound file (Reported by Tzafrir Cohen)
* ASTERISK-26739 - voicemail API test: confuses expected and
actual values (Reported by Tzafrir Cohen)
* ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
sorcery memory cache populate (Reported by Ustinov Artem)
* ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
(rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
(Reported by Aaron An)
* ASTERISK-26672 - Crash when setting remote address on RTP
instance (Reported by Richard Mudgett)
* ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
(Reported by Alexander Traud)
* ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
(Reported by Alexander Traud)
* ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
function around masquerade (Reported by Joshua Colp)
* ASTERISK-26684 - res_pjsip: Various issues with compact SIP
headers (Reported by Joshua Elson)
* ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
Headers Enabled (Reported by JoshE)
* ASTERISK-26621 - app_queue: Queue application does not ring
members with Local interface (Reported by Jonas Kellens)
* ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
MWI wasn't registered (Reported by Michael Kuron)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-26653 - pjproject_bundled doesn't verify already
downloaded tarballs (Reported by George Joseph)
* ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
setting up new calls (Reported by Walter Doekes)
* ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
line (Reported by Jørgen H)
* ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
aors (Reported by George Joseph)
* ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not
Exist when transaction branch parameter contains "_" (Reported
by Juris Breicis)
* ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without
IPv6 (Reported by Guido Falsi)
* ASTERISK-24330 - Requirement for 'wss' value in Contact header
transport parameter on inbound traffic violates RFC7118
(Reported by Marek Cervenka)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen)
* ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP
(Reported by Hector Royo Concepcion)
* ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to
tlscertfile, tlsciphers, etc. (Reported by Michael Kuron)
* ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec
to receiving codec when asymmetric_rtp_codec=no (Reported by
Alexei Gradinari)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
Improvements made in this release:
-----------------------------------
* ASTERISK-23828 - pjsip - Need a command to list active SIP
subscriptions (Reported by Rusty Newton)
* ASTERISK-26527 - Testsuite: increase timeout to check "core
fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav)
* ASTERISK-26624 - res_calendar_caldav: Add support for gmail
(Reported by Eduardo Scudeller Libardi)
* ASTERISK-26562 - app_controlplayback: Transmit Silence on
ControlPlayback pause (Reported by Mikheili Dautashvili)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0
Thank you for your continued support of Asterisk!
-----
Revision 1.25 / (download) - annotate - [select for diffs], Sun Apr 30 01:21:30 2017 UTC (6 years, 1 month ago) by ryoon
Branch: MAIN
Changes since 1.24: +2 -2
lines
Diff to previous 1.24 (colored)
Recursive revbump from boost update
Revision 1.24 / (download) - annotate - [select for diffs], Sat Apr 22 21:03:26 2017 UTC (6 years, 1 month ago) by adam
Branch: MAIN
Changes since 1.23: +2 -2
lines
Diff to previous 1.23 (colored)
Revbump after icu update
Revision 1.23 / (download) - annotate - [select for diffs], Sun Feb 12 06:25:09 2017 UTC (6 years, 3 months ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2017Q1-base,
pkgsrc-2017Q1
Changes since 1.22: +2 -2
lines
Diff to previous 1.22 (colored)
Recursive revbump from fonts/harfbuzz
Revision 1.22 / (download) - annotate - [select for diffs], Mon Feb 6 13:55:09 2017 UTC (6 years, 4 months ago) by wiz
Branch: MAIN
Changes since 1.21: +2 -2
lines
Diff to previous 1.21 (colored)
Recursive bump for harfbuzz's new graphite2 dependency.
Revision 1.21 / (download) - annotate - [select for diffs], Thu Jan 19 18:52:04 2017 UTC (6 years, 4 months ago) by agc
Branch: MAIN
Changes since 1.20: +4 -4
lines
Diff to previous 1.20 (colored)
Convert all occurrences (353 by my count) of MASTER_SITES= site1 \ site2 style continuation lines to be simple repeated MASTER_SITES+= site1 MASTER_SITES+= site2 lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint accordingly.
Revision 1.20 / (download) - annotate - [select for diffs], Sun Jan 1 16:06:06 2017 UTC (6 years, 5 months ago) by adam
Branch: MAIN
Changes since 1.19: +2 -2
lines
Diff to previous 1.19 (colored)
Revbump after boost update
Revision 1.19 / (download) - annotate - [select for diffs], Sun Dec 4 05:17:19 2016 UTC (6 years, 6 months ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2016Q4-base,
pkgsrc-2016Q4
Changes since 1.18: +2 -1
lines
Diff to previous 1.18 (colored)
Recursive revbump from textproc/icu 58.1
Revision 1.18 / (download) - annotate - [select for diffs], Sun Nov 27 08:48:18 2016 UTC (6 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.17: +2 -2
lines
Diff to previous 1.17 (colored)
Update to Asterisk 13.13.0: this is mainly a bug fix release with some
minor improvements.
The Asterisk Development Team has announced the release of Asterisk 13.13.0.
The release of Asterisk 13.13.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26595 - ARI: Add the ability to control the source of
video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
events (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
by snuffy)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content
(Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus
negotiated but codec_opus not loaded. (Reported by Richard
Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory
leak. (Reported by Richard Mudgett)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality
when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George
Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when
building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26468 - ari: Bridge events stop working after this
sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
payload types. (Reported by Alexander Traud)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state (Reported by Joshua
Colp)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
to maximum (Reported by Joshua Colp)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
reg. retry 403" in "sip show settings" (Reported by Sergey
Grachev)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented
(Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space
when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even
with no active calls. (Reported by Harley Peters)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport
in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components
option of tar which isn't supported in older versions (Reported
by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains
hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
(Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
installations. (Reported by Alexander Traud)
* ASTERISK-26421 - Segmentation Fault with ARI originate into
mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return
prompt. (Reported by John Kiniston)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already
disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
Tzafrir Cohen)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
Kayode)
* ASTERISK-26462 - [patch] app_queue: While using queues with
realtime, setting back to an empty context doesn't stop the exit
key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
Improvements made in this release:
-----------------------------------
* ASTERISK-25063 - [patch]add X.509 subject alternative name
support to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-26558 - app_queue: add variable to know if the call is
not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
(Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to
configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
blacklisting host subnets that are not involved in RTP (Reported
by Michael Walton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0
Thank you for your continued support of Asterisk!
Revision 1.17 / (download) - annotate - [select for diffs], Fri Nov 11 15:44:16 2016 UTC (6 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.16: +2 -2
lines
Diff to previous 1.16 (colored)
Update the Asterisk 13.12.2: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.2.
The release of Asterisk 13.12.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2
Thank you for your continued support of Asterisk!
Revision 1.16 / (download) - annotate - [select for diffs], Sat Oct 29 02:10:06 2016 UTC (6 years, 7 months ago) by jnemeth
Branch: MAIN
Changes since 1.15: +2 -2
lines
Diff to previous 1.15 (colored)
Update to Asterisk 13.12.1: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.1.
The release of Asterisk 13.12.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1
Thank you for your continued support of Asterisk!
Revision 1.15 / (download) - annotate - [select for diffs], Thu Oct 27 01:08:17 2016 UTC (6 years, 7 months ago) by jnemeth
Branch: MAIN
Changes since 1.14: +2 -3
lines
Diff to previous 1.14 (colored)
Update to Asterisk 13.12.0: this is mostly a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.0.
The release of Asterisk 13.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26277 - Add dialplan function
PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
update formats on a channel after session establishment
(Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26477 - pjproject: SEGV during SSL operations
(Reported by George Joseph)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk
allows one end peer to send video, even though the other end
supports only audio. (Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check for
all required utilities (Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that
may be flawed so we don't drop events (Reported by Richard
Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported
by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the
hangupcause on unanswered channels (Reported by George Joseph)
* ASTERISK-24311 - Populating database via Alembic fails when
using same database for multiple schema sets (Reported by Dafi
Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported
by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
bindaddr=[::] (Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
do_monitor lock (Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes Asterisk
14 (Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
Joshua Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
instructed (Reported by Tzafrir Cohen)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
(Reported by Anthony Messina)
* ASTERISK-26263 - SQL error when using realtime and registering
extension / inserting into ps_contacts (Reported by Jeppe Ryskov
Larsen)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
for connectionful protocols (Reported by Joshua Colp)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is
across multiple RTP packets (Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message
states seconds, but time value is milliseconds (Reported by
Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
by Aaron Hamstra)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to
extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
unexpected callerid (Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request
'REGISTER' failed (Reported by Dmitry Melekhov)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
(Reported by Etienne Lessard)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from
non-existent endpoint (Reported by nappsoft)
* ASTERISK-26288 - followme: fails to reset config items to
default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
argument) is enabled and callee rejects a call or hangs up.
(Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on "core show channeltype Surrogate" in
ast_format_cap_get_names (Reported by CGI.NET)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro
"lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action
"ShowDialplan" when there's a circular dependency between
contexts (Reported by Etienne Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian
6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26299 - app_queue: Queue application sometimes stops
calling members with Local interface (Reported by Etienne
Lessard)
* ASTERISK-26203 - res_fax: Deadlock when using
FAXOPT(gateway)=yes with Local channels (Reported by Etienne
Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
inversion in T.38 query option with features bridging code
(Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
local channels. (Reported by Richard Mudgett)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup (Reported by nappsoft)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to
res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
abelbeck)
* ASTERISK-25472 - Swagger scripts are not replacing format
variable in file brief (Reported by Corey Farrell)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive
(Reported by Joshua Colp)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension
names (Reported by Corey Farrell)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
chan_pjsip (Reported by Ross Beer)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding
dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on
failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid
and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option
is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user
causes crash (Reported by Joshua Colp)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a
member with a forward to an nonexistent extension (Reported by
Etienne Lessard)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version
1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26280 - DNS lookups can block channel media paths
(Reported by Mark Michelson)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
similar treatment for module unloading as
res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system
initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on
DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..." (Reported by Hans
van Eijsden)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the
translation codec. (Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause
inefficiencies (Reported by Mark Michelson)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
dialplan know what fax transport was used (Reported by Alexei
Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0
Thank you for your continued support of Asterisk!
Revision 1.14 / (download) - annotate - [select for diffs], Sun Oct 9 21:41:55 2016 UTC (6 years, 8 months ago) by wiz
Branch: MAIN
Changes since 1.13: +2 -2
lines
Diff to previous 1.13 (colored)
Recursive bump for all users of pgsql now that the default is 95.
Revision 1.13 / (download) - annotate - [select for diffs], Fri Oct 7 18:25:40 2016 UTC (6 years, 8 months ago) by adam
Branch: MAIN
Changes since 1.12: +2 -1
lines
Diff to previous 1.12 (colored)
Revbump post boost update
Revision 1.12 / (download) - annotate - [select for diffs], Fri Sep 23 17:50:19 2016 UTC (6 years, 8 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2016Q3-base,
pkgsrc-2016Q3
Changes since 1.11: +9 -3
lines
Diff to previous 1.11 (colored)
Update to Asterisk 13.11.2: this is mainly a bug fix release
including two security issues: AST-2016-006 and AST-2016-007.
Note that AST-2016-006 only affected setups using PJSIP, which
pkgsrc Asterisk does not.
pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminte conflict with new hmac(1) function on NetBSD
----- AST-2016-006
Asterisk can be crashed remotely by sending an ACK to it from an
endpoint username that Asterisk does not recognize. Most SIP
request types result in an "artificial" endpoint being looked up,
but ACKs bypass this lookup. The resulting NULL pointer results in
a crash when attempting to determine if ACLs should be applied.
This issue was introduced in the Asterisk 13.10 release and only
affects that release.
This issue only affects users using the PJSIP stack with Asterisk.
Those users that use chan_sip are unaffected.
----- AST-2016-007
The overlap dialing feature in chan_sip allows chan_sip to report
to a device that the number that has been dialed is incomplete and
more digits are required. If this functionality is used with a
device that has performed username/password authentication RTP
resources are leaked. This occurs because the code fails to release
the old RTP resources before allocating new ones in this scenario.
If all resources are used then RTP port exhaustion will occur and
no RTP sessions are able to be set up.
----- 13.11.2
The Asterisk Development Team has announced the release of Asterisk 13.11.2.
The release of Asterisk 13.11.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request
'REGISTER' failed (Reported by Dmitry Melekhov)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2
Thank you for your continued support of Asterisk!
----- 13.11.0
The Asterisk Development Team has announced the release of Asterisk 13.11.0.
The release of Asterisk 13.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup (Reported by nappsoft)
* ASTERISK-26299 - app_queue: Queue application sometimes stops
calling members with Local interface (Reported by Etienne
Lessard)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..." (Reported by Hans
van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by
Richard Mudgett)
* ASTERISK-26227 - sqlalchemy error due to long identifier name
(Reported by Mark Michelson)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
by Aaron Hamstra)
* ASTERISK-26214 - Allow arbitrary time for fax detection to end
on a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling (Reported by Ben
Smithurst)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
executing Playback (Reported by Richard Mudgett)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
DTD in docs. (Reported by Alexander Traud)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
conditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
number even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime
performance - remove unneeded check on endpoint's contacts.
(Reported by Alexei Gradinari)
* ASTERISK-26133 - app_queue: Queue members receive multiple calls
(Reported by Richard Miller)
* ASTERISK-26196 - pbx: Time based includes can leak timezone
string (Reported by Corey Farrell)
* ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
(Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
DTLS failure occurred on RTP instance (Reported by Edwin
Vandamme)
* ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
ast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-25289 - Build System does not respect CFLAGS and
CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
of bounds and bugs (Reported by Alexei Gradinari)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-26184 - chan_sip: Reference leaks in error paths.
(Reported by Corey Farrell)
* ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
during duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
reuse (Reported by Scott Griepentrog)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
by Joshua Colp)
* ASTERISK-26172 - res_sorcery_realtime: fix bug when successful
sql UPDATE is treated as failed if there is no affected rows.
(Reported by Alexei Gradinari)
* ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
(Reported by Dmitriy Serov)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
Alexei Gradinari)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)
* ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
(Reported by Daniel Denson)
* ASTERISK-26326 - Crash when dialing MulticastRTP channel
(Reported by George Joseph)
Improvements made in this release:
-----------------------------------
* ASTERISK-26220 - Add support for noreturn function attributes.
(Reported by Corey Farrell)
* ASTERISK-22131 - Update the make dependencies script to pull,
build, and install the correct pjproject (Reported by Matt
Jordan)
* ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
(Reported by JoshE)
* ASTERISK-26159 - res_hep: enabled by default and information
sent to default address (Reported by Ross Beer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0
Thank you for your continued support of Asterisk!
Revision 1.11 / (download) - annotate - [select for diffs], Wed Aug 3 10:22:35 2016 UTC (6 years, 10 months ago) by adam
Branch: MAIN
Changes since 1.10: +2 -1
lines
Diff to previous 1.10 (colored)
Revbump after graphics/gd update
Revision 1.10 / (download) - annotate - [select for diffs], Sun Jul 24 06:35:50 2016 UTC (6 years, 10 months ago) by jnemeth
Branch: MAIN
Changes since 1.9: +2 -3
lines
Diff to previous 1.9 (colored)
Update to Asterisk 13.10.0: this is mainly a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
The release of Asterisk 13.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
"call_id" to contacts (Reported by Alexei Gradinari)
* ASTERISK-25994 - [patch]res_pjsip: module load priority
(Reported by Alexei Gradinari)
* ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
by Alexei Gradinari)
* ASTERISK-25835 - Authentication using 'Username' field from
Digest (Reported by Ross Beer)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
performace (Reported by Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
v21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
self-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
generates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
Michelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
pjproject isn't installed in a system location (Reported by
George Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* ASTERISK-26127 - res_pjsip_session: Crash due to race condition
between res_pjsip_session unload and timer (Reported by Joshua
Colp)
* ASTERISK-26083 - ARI: Announcer channels staying around after
playback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
http.conf (Reported by Alexander Traud)
* ASTERISK-26069 - Asterisk truncates To: header, dropping the
closing '>' (Reported by Vasil Kolev)
* ASTERISK-26097 - [patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)
* ASTERISK-25262 - Memory leak when a caller channel does multiple
dials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
Remotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26096 - res_hep: Crash when configuration file is
missing (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIP
Realtime (Reported by Scott Griepentrog)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
Davis)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
cr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel without
an originator adds all audio formats to it's capabilities
(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported by
Etienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
properly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
documentation needs clarification for when read/write is
possible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by Badalian
Vyacheslav)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26029 - parking: ast_parking_park_call should return
parking_space instead of parking_exten (Reported by Diederik de
Groot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
response (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
fields (Reported by Joshua Colp)
* ASTERISK-24986 - keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to
early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires
header in re-invite after proxy authentication is required
(Reported by George Joseph)
* ASTERISK-25964 - Outbound registrations created via ARI/push
configuration do not clean up outbound registrations currently
in flight (Reported by Matt Jordan)
* ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
into 1 TCP packet (Reported by Ross Beer)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different then
res_hep (Reported by Kevin Scott Adams)
* ASTERISK-26008 - app_followme does not delete recorded name
prompt (Reported by Tzafrir Cohen)
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
* ASTERISK-25990 - PJSIP TLS registration should respect
client_uri scheme when generating Contact URI (Reported by
Sebastian Damm)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
source port in nonce verification (Reported by Mark Michelson)
* ASTERISK-25993 - pjproject: Allow bundling to not require
everything it does (Reported by Joshua Colp)
* ASTERISK-25956 - Compilation error in conditionally compiled
code in config_options.c (Reported by Chris Trobridge)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported
by Joshua Colp)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-25968 - pjproject_bundled: Configure and make need to
be re-tested (Reported by George Joseph)
* ASTERISK-24463 - Voicemail email address corrupt or not sent
when message is in the process of being recorded during reload
(Reported by John Campbell)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
when running test (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
events for autocreated peers (Reported by Kirill Katsnelson)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
are case sensitive to QueueName (Reported by Javier Acosta)
New Features made in this release:
-----------------------------------
* ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0
Thank you for your continued support of Asterisk!
Revision 1.9 / (download) - annotate - [select for diffs], Sat Jul 9 06:37:55 2016 UTC (6 years, 11 months ago) by wiz
Branch: MAIN
Changes since 1.8: +2 -1
lines
Diff to previous 1.8 (colored)
Bump PKGREVISION for perl-5.24.0 for everything mentioning perl.
Revision 1.8 / (download) - annotate - [select for diffs], Thu Jun 9 04:41:48 2016 UTC (7 years ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2016Q2-base,
pkgsrc-2016Q2
Changes since 1.7: +2 -2
lines
Diff to previous 1.7 (colored)
Upgrade to Asterisk 13.9.1: this is a bugfix release. Note that
since the package doesn't support PJSIP (yet), all reference to
PJSIP bugs are not applicable.
----- 13.9.1
The Asterisk Development Team has announced the release of Asterisk 13.9.1.
The release of Asterisk 13.9.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1
Thank you for your continued support of Asterisk!
----- 13.9.0
The Asterisk Development Team has announced the release of Asterisk 13.9.0.
The release of Asterisk 13.9.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25927 - Removed option "registertrying" is still
documented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications are
missing the StasisStart with the replace_channel object.
(Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs or
updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash
pjproject/Asterisk under certain conditions (Reported by George
Joseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter
unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
without adding them to the local hangupcauses via
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between adding
contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-25910 - pjproject: Via headers are not parsed when
"received" contains an IPv6 address (Reported by George Joseph)
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
(Reported by Harley Peters)
* ASTERISK-25894 - [patch] webrtc video broken due to missing
marker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
cannot find -lasteriskpj (Reported by Hans van Eijsden)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Part 2) (Reported by
Richard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
Jacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not work
with the core Features 'parkcall' (DTMF initiated parking)
(Reported by Philip Correia)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-24596 - Unclear how to use Park application with
res_parking 'parkeddynamic' enabled. Documentation? (Reported by
Philip Correia)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25825 - Crashes during shutdown when running CLI
commands (Reported by Mark Michelson)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by
Michael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
data corruption (Reported by Gianluca Merlo)
Improvements made in this release:
-----------------------------------
* ASTERISK-25865 - Message-Account Missing From PJSIP MWI
(Reported by Ross Beer)
* ASTERISK-25444 - [patch]Music On Hold Warning misleading
(Reported by Conrad de Wet)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0
Thank you for your continued support of Asterisk!
Revision 1.7 / (download) - annotate - [select for diffs], Fri May 6 07:41:06 2016 UTC (7 years, 1 month ago) by jnemeth
Branch: MAIN
Changes since 1.6: +9 -9
lines
Diff to previous 1.6 (colored)
Update to Asterisk 13.8.2: this is mainly a bug fix release. It
also contains fixes for AST-2016-004 and AST-2016-005. However,
those issues only affected the pjsip module. Since Asterisk in
pkgsrc doesn't (yet) use pjsip, it wasn't affected.
----- 13.8.2
The Asterisk Development Team has announced the release of Asterisk 13.8.2.
The release of Asterisk 13.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2
Thank you for your continued support of Asterisk!
----- 13.8.0
The Asterisk Development Team has announced the release of Asterisk 13.8.0.
The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
Journo)
* ASTERISK-25480 - [patch]Add field PauseReason on
QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25849 - chan_pjsip: transfers with direct media
sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard
system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in
update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache
(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
PJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
OutboundSubscriptionDetail ami action (Reported by Kevin
Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and
heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support
pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery
(Reported by Martin Mouka)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option
not in Alembic (Reported by Joshua Colp)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in
weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Reported by Mark
Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and
Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
test sporadically failing (Reported by Joshua Colp)
* ASTERISK-24097 - Documentation - CHANNEL function help text
missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer
fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by
Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25317 - asterisk sends too many stun requests (Reported
by Stefan Engström)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-25495 - [patch] Prevent old-update packages on
repository Debian systems (Reported by Rodrigo Ramirez
Norambuena)
* ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
(Reported by Andrew Nagy)
* ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
Messina)
* ASTERISK-24813 - asterisk.c: #if statement in listener()
confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
core set (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0
Thank you for your continued support of Asterisk!
Revision 1.6 / (download) - annotate - [select for diffs], Mon Apr 11 19:01:44 2016 UTC (7 years, 2 months ago) by ryoon
Branch: MAIN
Changes since 1.5: +2 -2
lines
Diff to previous 1.5 (colored)
Recursive revbump from textproc/icu 57.1
Revision 1.5 / (download) - annotate - [select for diffs], Sat Mar 5 11:28:10 2016 UTC (7 years, 3 months ago) by jperkin
Branch: MAIN
CVS Tags: pkgsrc-2016Q1-base,
pkgsrc-2016Q1
Changes since 1.4: +2 -1
lines
Diff to previous 1.4 (colored)
Bump PKGREVISION for security/openssl ABI bump.
Revision 1.4 / (download) - annotate - [select for diffs], Thu Feb 25 11:32:19 2016 UTC (7 years, 3 months ago) by jperkin
Branch: MAIN
Changes since 1.3: +3 -5
lines
Diff to previous 1.3 (colored)
Use OPSYSVARS.
Revision 1.3 / (download) - annotate - [select for diffs], Sun Feb 7 09:13:34 2016 UTC (7 years, 4 months ago) by jnemeth
Branch: MAIN
Changes since 1.2: +20 -18
lines
Diff to previous 1.2 (colored)
Update Asterisk to 13.7.2: this is mainly bug fixes with some minor
features and fixes for AST-2016-001, AST-2016-002, and AST-2016-003.
Also some pkglinting.
----- 13.7.2
The Asterisk Development Team has announced the release of Asterisk 13.7.2.
The release of Asterisk 13.7.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2
Thank you for your continued support of Asterisk!
----- 13.7.1
The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
The release of these versions resolves the following security vulnerabilities:
* AST-2016-001: BEAST vulnerability in HTTP server
The Asterisk HTTP server currently has a default configuration which allows
the BEAST vulnerability to be exploited if the TLS functionality is enabled.
This can allow a man-in-the-middle attack to decrypt data passing through it.
* AST-2016-002: File descriptor exhaustion in chan_sip
Setting the sip.conf timert1 value to a value higher than 1245 can cause an
integer overflow and result in large retransmit timeout times. These large
timeout values hold system file descriptors hostage and can cause the system
to run out of file descriptors.
* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
If no UDPTL packets are lost there is no problem. However, a lost packet
causes Asterisk to use the available error correcting redundancy packets. If
those redundancy packets have zero length then Asterisk uses an uninitialized
buffer pointer and length value which can cause invalid memory accesses later
when the packet is copied.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
Thank you for your continued support of Asterisk!
----- 13.7.0
The Asterisk Development Team has announced the release of Asterisk 13.7.0.
The release of Asterisk 13.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option
(Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25689 - pjsip show contacts not working in Asterisk
13.7rc2 (Reported by Marcelo Terres)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations >
1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25619 - res_chan_stats not sending the correct
information to StatsD (Reported by Tyler Cambron)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
Corey Farrell)
* ASTERISK-25609 - [patch]Asterisk may crash when calling
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when
answer waiting time is more than ~7sec (Reported by Aleksei
Kulakov)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC
with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events
aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without
that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
by Niklas Larsson)
* ASTERISK-25476 - chan_sip loses registrations after a while
(Reported by Michael Keuter)
* ASTERISK-25598 - res_pjsip: Contact status messages are
printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
by Jonathan Rose)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25593 - fastagi: record file closed after sending
result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
created via ARI are not loaded into memory on Asterisk
start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not
joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets
whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
'qe->chan' freed more times than we've locked! (Reported by Alec
Davis)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead
of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout
and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or
internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
Doekes)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with
chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI
originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show
settings' despite usage during compilation (Reported by Rusty
Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it
will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of
subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
dialog can't be created (Reported by Joshua Colp)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration
stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values
in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
by Alexander Traud)
* ASTERISK-25461 - Nested dialplan #includes don't work as
expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over
res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
(Reported by Bojan Nemi)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-25618 - res_pjsip: Check for readability of TLS files
at startup (Reported by George Joseph)
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
endpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
objects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
Jonathan Rose)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
by Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to
configure (Reported by Badalian Vyacheslav)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0
Thank you for your continued support of Asterisk!
Revision 1.2 / (download) - annotate - [select for diffs], Sat Dec 5 23:42:44 2015 UTC (7 years, 6 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2015Q4-base,
pkgsrc-2015Q4
Changes since 1.1: +1 -1
lines
Diff to previous 1.1 (colored)
Initial import of Asterisk 13. It has been tested to compile
and run, but not a lot of functional testing. This does not have
the new PJSIP, which will be coming in a followup commit. This
also does not have the patches for compiling with Clang. For
upgrading instructions, please see:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13
-----
The Asterisk Development Team is pleased to announce the release
of Asterisk 13.0.0.
Asterisk 13 is the next major release series of Asterisk. It is a
Long Term Support (LTS) release, similar to Asterisk 11. For more
information about support time lines for Asterisk releases, see
the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
For important information regarding upgrading to Asterisk 13, please
see the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13
A short list of new features includes:
* Asterisk security events are now provided via AMI, allowing end
users to monitor their Asterisk system in real time for security
related issues.
* Both AMI and ARI now allow external systems to control the state
of a mailbox. Using AMI actions or ARI resources, external
systems can programmatically trigger Message Waiting Indicators
(MWI) on subscribed phones. This is of particular use to those
who want to build their own VoiceMail application using ARI.
* ARI now supports the reception/transmission of out of call text
messages using any supported channel driver/protocol stack through
ARI. Users receive out of call text messages as JSON events over
the ARI websocket connection, and can send out of call text
messages using HTTP requests.
* The PJSIP stack now supports RFC 4662 Resource Lists, allowing
Asterisk to act as a Resource List Server. This includes defining
lists of presence state, mailbox state, or lists of presence
state/mailbox state; managing subscriptions to lists; and batched
delivery of NOTIFY requests to subscribers.
* The PJSIP stack can now be used as a means of distributing device
state or mailbox state via PUBLISH requests to other Asterisk
instances. This is analogous to Asterisk's clustering support
using XMPP or Corosync; unlike existing clustering mechanisms,
using the PJSIP stack to perform the distribution of state does
not rely on another daemon or server to perform the work.
And much more!
More information about the new features can be found on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation
A full list of all new features can also be found in the CHANGES file:
http://svnview.digium.com/svn/asterisk/branches/13/CHANGES
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.1.0.
The release of Asterisk 13.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24554 - AMI/ARI: Generate events on connected line
changes (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
Corey Farrell)
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
leak (Reported by Corey Farrell)
* ASTERISK-24430 - missing letter "p" in word response in
OriginateResponse event documentation (Reported by Dafi Ni)
* ASTERISK-24437 - Review implementation of ast_bridge_impart for
leaks and document proper usage (Reported by Scott Griepentrog)
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
Corey Farrell)
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
Corey Farrell)
* ASTERISK-24458 - chan_phone fails to build on big endian systems
(Reported by Tzafrir Cohen)
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
* ASTERISK-24304 - asterisk crashing randomly because of unistim
channel (Reported by dhanapathy sathya)
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
Nick Adams)
* ASTERISK-24462 - res_pjsip: Stale qualify statistics after
disablementation (Reported by Kevin Harwell)
* ASTERISK-24465 - audiohooks list leaks reference to formats
(Reported by Corey Farrell)
* ASTERISK-24466 - app_queue: fix a couple leaks to struct
call_queue (Reported by Corey Farrell)
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
(Reported by Corey Farrell)
* ASTERISK-24411 - [patch] Status of outbound registration is not
changed upon unregistering. (Reported by John Bigelow)
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
leaks (Reported by Corey Farrell)
* ASTERISK-24480 - res_http_websockets: Module reference decrease
below zero (Reported by Corey Farrell)
* ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
audiohook callback (Reported by Corey Farrell)
* ASTERISK-24487 - configuration: sections should be loadable as
template even when not marked (Reported by Scott Griepentrog)
* ASTERISK-20127 - [Regression] Config.c config_text_file_load()
unescapes semicolons ("\;" -> ";") turning them into comments
(corruption) on rewrite of a config file (Reported by George
Joseph)
* ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
when DNS settings invalid (Reported by Melissa Shepherd)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
* ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
Conkle)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
extra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
waiting for more matching digits. (Reported by Richard Mudgett)
* ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
queue caller (Reported by Steve Pitts)
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)
* ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
header fix (Reported by abelbeck)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
length exceeds 50 (roughly) national symbols (Reported by
Dmitriy Bubnov)
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
revision r227276 (Reported by Xavier Hienne)
* ASTERISK-24505 - manager: http connections leak references
(Reported by Corey Farrell)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
coverage are enabled (Reported by Corey Farrell)
* ASTERISK-24444 - PBX: Crash when generating extension for
pattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
packet to JSON for res_hep_rtcp and report blocks are greater
than 1 (Reported by Gregory Malsack)
* ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
transfer (Reported by Beppo Mazzucato)
* ASTERISK-24501 - ARI: Moving a channel between bridges followed
by a hangup can cause an ARI client to not receive an expected
ChannelLeftBridge event before StasisEnd (Reported by Matt
Jordan)
* ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
(Reported by Leon Rowland)
* ASTERISK-23651 - Reloading some modules that are loaded already,
results in 'No such module' before a successful reload (Reported
by Rusty Newton)
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
endmarked users when last marked user leaves (Reported by Matt
Jordan)
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
(Reported by David Woolley)
* ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
with "400 bad request" - DEBUG shows "Received a REFER without a
parseable Refer-To" (Reported by Beppo Mazzucato)
* ASTERISK-24535 - stringfields: Fix regression from fix for
unintentional memory retention and another issue exposed by the
fix (Reported by Corey Farrell)
* ASTERISK-24471 - Crash - assert_fail in libc in
pjmedia_sdp_neg_negotiateofrom /usr/local/lib/libpjmedia.so.2
(Reported by yaron nahum)
* ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
in-dialog with invalid target causes crash (Reported by Joshua
Colp)
* ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
module load (Reported by Matt Jordan)
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
allow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24542 - [patch]Failure showing codecs via 'core show
channeltype <tech>' (Reported by snuffy)
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
by xrobau)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24572 - [patch]App_meetme is loaded without its
defaults when the configuration file is missing (Reported by
Nuno Borges)
* ASTERISK-24573 - [patch]Out of sync conversation recording when
divided in multiple recordings (Reported by NunowBorges)
* ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
reliably transmitted during transfers (Reported by Matt Jordan)
* ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
extension to another pjsip extension (Reported by Abhay Gupta)
Improvements made in this release:
-----------------------------------
* ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
property 'unanswered' (Reported by Matt Jordan)
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
column in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
recordings (Reported by Ben Smithurst)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
lookups (Reported by Birger "WIMPy" Harzenetter)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.2.0.
The release of Asterisk 13.2.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
all at the same time. (Reported by Richard Mudgett)
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
when using non-default sorcery wizard (Reported by Kevin
Harwell)
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
media streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24563 - Direct Media calls within private network
sometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
race condition in accessing codec in stored ast_frame and codec
core (Reported by Matt Jordan)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
channel (Reported by Niklas Larsson)
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
chosen for RTP compatible channels when the DTMF mode is not
compatible (Reported by Yaniv Simhi)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
level - 'Remote address is null, most likely RTP has been
stopped' (Reported by Rusty Newton)
* ASTERISK-24513 - Local channel apparently leaked in off-nominal
DTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
destination when 'sendrpid=yes' (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
calls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24376 - res_pjsip_refer: REFER request for remote
session attempts to direct channel to external_replaces
extension instead of context, without providing for the
Referred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24591 - Stasis() side of an ARI originated channel
cannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24049 - Asterisk Manager Interface: A number of list
type responses aren't using astman_send_listack (Reported by
Jonathan Rose)
* ASTERISK-24637 - Channel re-enters Stasis() when it should not
(Reported by John Bigelow)
* ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
not function (Reported by John Kiniston)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)
* ASTERISK-20744 - [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-24665 - Configure check required for
pjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-23850 - Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
while attempting to publish (Reported by Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
on cross compilation (Reported by abelbeck)
* ASTERISK-24624 - Transfer to invalid extension results in hung
channel. (Reported by Zane Conkle)
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
Incorrect External Addresses is Used in SIP Packets When
Responding to INVITE (Reported by David Justl)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
to most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24560 - Creating a named ARI bridge twice causes a
crash (Reported by Kinsey Moore)
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
by Matt Jordan)
* ASTERISK-24640 - Registration pending stays forever after sip
reload (Reported by Max Man)
* ASTERISK-24673 - outgoing sip registers cannot be removed or
modified without doing restart (or doing module unload
chan_sip.so) (Reported by Stefan Engström)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
column comparison for 'defaultuser' (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)
* ASTERISK-24626 - Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because of
incorrect detection of htonll and ntohll (Reported by George
Joseph)
* ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
'module not found' during a Reload operation (Reported by Matt
Jordan)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24729 - Outbound registration not occuring on new
registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24666 - Security Vulnerability: RTP not closed after
sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
is ever received (Reported by Marco Paland)
* ASTERISK-24737 - When agent not logged in, agent status shows
unavailable, queue status shows agent invalid (Reported by
Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-24552 - ARI: Allow associating a channel as an
initiator of an Origination for record keeping purposes
(Reported by Matt Jordan)
* ASTERISK-24553 - ARI/AMI: Include language in standard channel
snapshot output (Reported by Matt Jordan)
* ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
Matt Jordan)
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
Israel))
* ASTERISK-24678 - [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
Dan Jenkins)
* ASTERISK-24316 - For httpd server, need option to define server
name for security purposes (Reported by Andrew Nagy)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.2.1.
The release of Asterisk 13.2.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
* --- pjsip: resolve compatibility problem with ast_sip_session
(Closes issue ASTERISK-24941. Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.1
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.3.0.
The release of Asterisk 13.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
channel (Reported by Matt Jordan)
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
string copy (Reported by Yura Kocyuba)
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
sorcery.conf false ERROR messages may occur (Reported by Joshua
Colp)
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
(Reported by Matt Jordan)
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
is destroyed by ARI during shutdown (Reported by Richard
Mudgett)
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
by Zane Conkle)
* ASTERISK-24015 - app_transfer fails with PJSIP channels
(Reported by Private Name)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
transfer scenario. (Reported by Mark Michelson)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
Niklas Larsson)
* ASTERISK-24716 - Improve pjsip log messages for presence
subscription failure (Reported by Rusty Newton)
* ASTERISK-24612 - res_pjsip: No information if a required sorcery
wizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak
(Reported by Matthias Urlichs)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by
Joshua Colp)
* ASTERISK-24632 - install_prereq script installs pjproject
without IPv6 support (Reported by Rusty Newton)
* ASTERISK-24085 - Documentation - We should remove or further
document the 'contact' section in pjsip.conf (Reported by Rusty
Newton)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
JoshE)
* ASTERISK-24700 - CRASH: NULL channel is being passed to
ast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
Events (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
call (Reported by Marcel Manz)
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
(Reported by Panos Gkikakis)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-24785 - 'Expires' header missing from 200 OK on
REGISTER (Reported by Ross Beer)
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful
response on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24812 - ARI: Creating channels through /channels
resource always uses SLIN, which results in unneeded transcoding
(Reported by Matt Jordan)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-24751 - Integer values in json payload to ARI cause
asterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
HAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
(Reported by Kevin Harwell)
* ASTERISK-24755 - Asterisk sends unexpected early BYE to
transferrer during attended transfer when using a Stasis bridge
(Reported by John Bigelow)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
by Anatoli)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
connection on error (Reported by Dmitriy Serov)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
by Corey Farrell)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
Improvements made in this release:
-----------------------------------
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
(Reported by Ben Merrills)
* ASTERISK-24811 - asterisk-publication sorcery object does not
use realtime (Reported by Matt Hoskins)
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
Couldn't find mailbox %s in context (Reported by Graham Barnett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.3.1.
The release of Asterisk 13.3.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
* --- pjsip: resolve compatibility problem with ast_sip_seesion
(Closes issue ASTERISK-24941. Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.4.0.
The release of Asterisk 13.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25112 - Logger: Configuration settings are not reset to
default during reload. (Reported by Corey Farrell)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call
recording (Reported by Ronald Raikes)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
templates aren't being processed correctly (Reported by George
Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables
(Reported by snuffy)
* ASTERISK-25085 - [patch]Potential crash after unload of
func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-25082 - Asterisk deletes message after doing a playback
of an INBOX message using ast_vm_play when the Old folder is
full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-25041 - [patch]Broken column type checking in
res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in
ast_channel_hangupcause_set, at channel_internal_api.c (Reported
by Alexandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke
cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line
options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when
set in the future (Reported by tootai)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive
escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using
feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing
dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with
module version of macro. (Reported by Corey Farrell)
* ASTERISK-25025 - Periodic crashes (in
ast_channel_snapshot_create at stasis_channels.c) with Certified
Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing
trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option
not respected, failover between DSNs doesn't work (Reported by
JoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered,
needs to hold modules until shutdown. (Reported by Corey
Farrell)
* ASTERISK-24896 - [patch] Using force black background leads to
colours not being reset (Reported by dant)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
PJSip (Reported by Peter Whisker)
* ASTERISK-25028 - Build System: Unneeded defines in
asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25048 - Astobj2: Initialization order wrong when both
refdebug and AO2_DEBUG are both enabled. (Reported by Corey
Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with
cause code 44 after some time. (Reported by Denis Alberto
Martinez)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potential
crash in off-nominal failure case when sending message (Reported
by Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
which is disallowed in res_fax's check_modem_rate (Reported by
Matt Jordan)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
by Ashley Sanders)
* ASTERISK-25020 - Mismatched response to outgoing REGISTER
request (Reported by Mark Michelson)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk when
qualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playing
conf-hasjoin and conf-hasleft when played to bridge (Reported by
Philippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone
(Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite after
originated T.38 FAX (Reported by Mark Michelson)
* ASTERISK-24999 - PJSIP crashes with malformed contact line
(Reported by snuffy)
* ASTERISK-24998 - res_corosync: res_corosync tries to load even
if res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
pre-check the object (Reported by Corey Farrell)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
on mailbox changes (Reported by Joshua Colp)
* ASTERISK-24991 - Check for ao2_alloc failure in
__ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no
HangupRequest event comes for the dahdi channel. (Reported by
Andrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are being
marked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy with
extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
to Fail (Reported by Ashley Sanders)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
when contacts cannot be reached/qualified (Reported by Dmitriy
Serov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
due to application (appl) being NULL on unbridged channel
(Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
notify (Reported by Scott Griepentrog)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24954 - Git migration: Asterisk version numbers are
incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
openssl not compiled (Reported by Warren Selby)
* ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
honored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP and
Asterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead of
audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with
certain audio/video codec configuration, resulting in path
translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
short (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)
* ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
Rose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of
iaxs[peer->callno] potentially results in segfault (Reported by
Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-24910 - "timer=no" and "timer=required" settings in
pjsip.conf fail (Reported by Ray Crumrine)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)
* ASTERISK-24914 - Division by zero in file.c when playback of
voicemail with video as h264 (Reported by Marcello Ceschia)
* ASTERISK-24899 - Parking fall-through behavior different in 13
(Reported by Malcolm Davenport)
* ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
sent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
they were each a new request (Reported by Mark Michelson)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
calls, voicemail prompts and recordings all fail when using the
kqueue timer source on FreeBSD 10.x (Reported by Justin T.
Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device
lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition
(heap-use-after-free) on asterisk closing (Reported by Badalian
Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when
absolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when
processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered
when processing the roster (Reported by Simon Arlott)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
with undesireabe consequences. (Reported by Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-25044 - sorcery: Add ability to insert a new wizard
into an object type's list (Reported by George Joseph)
* ASTERISK-24892 - Super Awesome Company sound prompts (Reported
by Rusty Newton)
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
Hjelm)
* ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
(Reported by Alexander Traud)
* ASTERISK-25045 - vector: Add new capabilities and unit tests
(Reported by George Joseph)
* ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
by yaron nahum)
* ASTERISK-25051 - Remove unneeded uses of optional_api providers.
(Reported by Corey Farrell)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported by
Diederik de Groot)
* ASTERISK-24949 - res_pjsip_outbound_registration: Backport line
functionality (Reported by Joshua Colp)
* ASTERISK-24965 - cel_pgsql - log_error string references CDR
instead of CEL (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24918 - pjsip: add CLI options to display global and
system configuration (Reported by Scott Griepentrog)
* ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
yaron nahum)
* ASTERISK-24802 - stasis: set a channel variable on websocket
disconnect error (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.5.0.
The release of Asterisk 13.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
when Asterisk deletes a dialplan variable. (Reported by Richard
Mudgett)
* ASTERISK-25067 - Sorcery Caching: Implement a new caching module
(Reported by Matt Jordan)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-25114 - res_pjsip: Add AMI etents for chan_pjsip
contact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: line
functionality. Additional check for using the request URI
(Reported by Dmitriy Serov)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume
controls such as func_volume don't work (Reported by Dmitriy
Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
Newton)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
endpoints outside NAT - implement functionality similar to
chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
RTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escape
control characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to an
empty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
empty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel in
chan_pjsip_incoming_response despite previous checks for NULL
channel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'd
threadpool (Reported by Matt Jordan)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
started when completing attended transfer (Reported by Joshua
Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel
schedule ID" in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
ast_channel_name at channel_internal_api.c (Reported by Carl
Fortin)
* ASTERISK-25115 - Crash related to func
sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
(Reported by John Bigelow)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
replaces call pickup (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
applied to Contact header when Record-Route headers are present
(Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash during
unload if registration attempts are still occuring (Reported by
Kevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
Replaces headers on outbound INVITEs. (Reported by Mark
Michelson)
* ASTERISK-25171 - Early completion of feature code attended
transfer results in intermittent one-way audio, "ghost ringing"
and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channel
snapshot information. (Reported by Richard Mudgett)
* ASTERISK-25172 - Crash in channels/sip/sip blind
transfer/caller_refer_only test in
ast_format_cap_append_from_cap during ast_request (Reported by
Matt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
(Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
appended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
container and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
asterisk when calling channel hangup while adding to bridge
(Reported by Ilya Trikoz)
* ASTERISK-24900 - Manager event ParkedCallSwap is not documented
(Reported by Rusty Newton)
* ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
(Reported by Corey Farrell)
* ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
negotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
dialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
call started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updatep after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsipÑÔ CHAN_START cel event lacks the
correct context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does not
result in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directory
application with alias option and specific mailbox configuration
(Reported by Chet Stevens)
* ASTERISK-24983 - IAX deadlock between hangup and scheduled
actions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-25096 - [patch]Segfault when registering over
websockets with PJSIP (in ast_sockaddr_isnull at
/include/asterisk/netsock2.h) (Reported by Josh Kitchens)
* ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
(Reported by Badalian Vyacheslav)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
Michelson)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25131 - chan_pjsip: In-dialog authentication not
handled. (Reported by Richard Mudgett)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip over
websocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
modules. (Reported by Corey Farrell)
* ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
(Reported by Joshua Colp)
* ASTERISK-25105 - res_pjsip: Possible incompatibility between
qualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager action
registrations. (Reported by Corey Farrell)
New Features made in this release:
-----------------------------------
* ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
Joshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported by
Matt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
Asterisk module (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.5.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 13.6.0.
The release of Asterisk 13.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
to something more palatable (Reported by Mark Michelson)
* ASTERISK-25252 - ARI: Add the ability to manipulate log channels
(Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25383 - Core dumps on startup and shutdown with
MALLOC_DEBUG enabled (Reported by yaron nahum)
* ASTERISK-25423 - Caller gets no Connected line update during
call pickup. (Reported by Richard Mudgett)
* ASTERISK-25305 - Dynamic logger channels can be added multiple
times (Reported by Mark Michelson)
* ASTERISK-25418 - On-hold channels redirected out of a bridge
appear to still be on hold (Reported by Mark Michelson)
* ASTERISK-25384 - Regular Asterisk crashes when using Page
application. "user_data is NULL" (Reported by Chet Stevens)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being
populated (Reported by Kevin Harwell)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes
invalid SIP (Reported by Walter Doekes)
* ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
(Reported by Kevin Harwell)
* ASTERISK-25185 - Segfault in app_queue on transfer scenarios
(Reported by Etienne Lessard)
* ASTERISK-25353 - [patch] Transcoding while different in Frame
size = Frames lost (Reported by Alexander Traud)
* ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25390 - default_from_user can crash with certain
configuration backends (Reported by Mark Michelson)
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
causes NAT'd Contact header to not be rewritten (Reported by
Matt Jordan)
* ASTERISK-25227 - No audio at in-band announcements in ooh323
channel (Reported by Alexandr Dranchuk)
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
variables aren't applied to the announcer channel (Reported by
Jonathan Rose)
* ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
/usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
* ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
mechanism) do not destroy their related contacts (Reported by
Matt Jordan)
* ASTERISK-25367 - pbx: Long pattern match hints may cause "core
show hints" to crash (Reported by Joshua Colp)
* ASTERISK-25365 - Persistent subscriptions have extra
Content-Length/corrupted messages (Reported by Mark Michelson)
* ASTERISK-25362 - Deadlock due to presence state callback
(Reported by Mark Michelson)
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
items may exist (Reported by Joshua Colp)
* ASTERISK-25355 - sched: ast_sched_del may return prematurely due
to spurious wakeup (Reported by Joshua Colp)
* ASTERISK-25318 -
tests/rest_api/applications/subscribe-endpoint/nominal/resource:
Sporadically failing (Reported by Joshua Colp)
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
cause on call pickup (Reported by Joshua Colp)
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
block (Reported by Joshua Colp)
* ASTERISK-25341 - bridge: Hangups may get lost when executing
actions (Reported by Joshua Colp)
* ASTERISK-25339 - res_pjsip: Empty "auth" sections from
non-config backgrounds are interpreted as valid (Reported by
Matt Jordan)
* ASTERISK-25215 - Differences in queue.log between Set
QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
Gaetz)
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
r() options. (Reported by Richard Mudgett)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
for wrong or non existent peer on invite (Reported by Kevin
Harwell)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF
tones. (Reported by Richard Mudgett)
* ASTERISK-25312 - res_http_websocket: Terminate connection on
fatal cases (Reported by Joshua Colp)
* ASTERISK-25306 - Persistent subscriptions can save multiple SIP
messages at once, leading to potential crashes. (Reported by
Mark Michelson)
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
Alexander Traud)
* ASTERISK-25304 - res_pjsip: XML sanitization may write past
buffer (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
Firefox 39 - add ECDH support and fallback to prime256v1
(Reported by Stefan Engström)
* ASTERISK-25296 - RTP performance issue with several channel
drivers. (Reported by Richard Mudgett)
* ASTERISK-25297 - Crashes running
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
(Reported by Richard Mudgett)
* ASTERISK-25292 - Testuite:
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
(Reported by Kevin Harwell)
* ASTERISK-25271 - Parking & blind transfer: Transferer channel
not hung up if no MOH (Reported by Kevin Harwell)
Improvements made in this release:
-----------------------------------
* ASTERISK-24870 - ARI: Subscriptions to bridges generally not
super useful (Reported by Matt Jordan)
* ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0
Thank you for your continued support of Asterisk!
Revision 1.1.1.1 / (download) - annotate - [select for diffs] (vendor branch), Sat Dec 5 23:29:05 2015 UTC (7 years, 6 months ago) by jnemeth
Branch: TNF
CVS Tags: pkgsrc-base
Changes since 1.1: +0 -0
lines
Diff to previous 1.1 (colored)
`cat ../comment`
Revision 1.1 / (download) - annotate - [select for diffs], Sat Dec 5 23:29:05 2015 UTC (7 years, 6 months ago) by jnemeth
Branch: MAIN
Initial revision