kpriv/pkgsrc
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
pkgsrc/comms/asterisk/patches/patch-main_pbx.c

32 lines
1.2 KiB
C
Raw Normal View History

Update Asterisk to 11.20.0: this is mainly a bug fix release. pkgsrc changes: - from joerg@ - srtp support - new asterisk-config option to control installing of sample config files - manifest.xml for Solaris' SMF - various bugfixes, some reworked by myself - backport kqueue timer update from Asterisk 13 ----- The Asterisk Development Team has announced the release of Asterisk 11.20.0. The release of Asterisk 11.20.0 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-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-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * 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-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk) * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp) * ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz) * 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-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström) Improvements made in this release: ----------------------------------- * 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-11.20.0 Thank you for your continued support of Asterisk!
2015-10-27 09:49:01 +01:00
$NetBSD: patch-main_pbx.c,v 1.2 2015/10/27 08:49:01 jnemeth Exp $
Update to Asterisk 11.17.1: this contains a security fix, plus various bugs. pkgsrc changes: - adapt to upstream support for clang - more comprehensive sweep for 64-bit time_t related stuff - XXX pjsip has its own time related stuff that is 32-bit only ----- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. The release of these versions resolves the following security vulnerability: * AST-2015-003: TLS Certificate Common name NULL byte exploit When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack. For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 The security advisory is available at: * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.17.0. The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE with replaces (Reported by Eelco Brolman) * 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-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-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-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * 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-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-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-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-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-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-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-11.17.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.16.0. The release of Asterisk 11.16.0 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-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * 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-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-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-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-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-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-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * 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-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0 Thank you for your continued support of Asterisk!
2015-05-19 09:52:14 +02:00
Update Asterisk to 11.20.0: this is mainly a bug fix release. pkgsrc changes: - from joerg@ - srtp support - new asterisk-config option to control installing of sample config files - manifest.xml for Solaris' SMF - various bugfixes, some reworked by myself - backport kqueue timer update from Asterisk 13 ----- The Asterisk Development Team has announced the release of Asterisk 11.20.0. The release of Asterisk 11.20.0 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-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-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * 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-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk) * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp) * ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz) * 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-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström) Improvements made in this release: ----------------------------------- * 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-11.20.0 Thank you for your continued support of Asterisk!
2015-10-27 09:49:01 +01:00
--- main/pbx.c.orig 2015-10-09 22:23:39.000000000 +0000
Update to Asterisk 11.17.1: this contains a security fix, plus various bugs. pkgsrc changes: - adapt to upstream support for clang - more comprehensive sweep for 64-bit time_t related stuff - XXX pjsip has its own time related stuff that is 32-bit only ----- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. The release of these versions resolves the following security vulnerability: * AST-2015-003: TLS Certificate Common name NULL byte exploit When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack. For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 The security advisory is available at: * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.17.0. The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE with replaces (Reported by Eelco Brolman) * 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-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-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-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * 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-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-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-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-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-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-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-11.17.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.16.0. The release of Asterisk 11.16.0 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-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * 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-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-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-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-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-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-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * 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-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0 Thank you for your continued support of Asterisk!
2015-05-19 09:52:14 +02:00
+++ main/pbx.c
Update Asterisk to 11.20.0: this is mainly a bug fix release. pkgsrc changes: - from joerg@ - srtp support - new asterisk-config option to control installing of sample config files - manifest.xml for Solaris' SMF - various bugfixes, some reworked by myself - backport kqueue timer update from Asterisk 13 ----- The Asterisk Development Team has announced the release of Asterisk 11.20.0. The release of Asterisk 11.20.0 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-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-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * 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-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk) * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp) * ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz) * 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-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström) Improvements made in this release: ----------------------------------- * 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-11.20.0 Thank you for your continued support of Asterisk!
2015-10-27 09:49:01 +01:00
@@ -11427,7 +11427,7 @@ static int testtime_write(struct ast_cha
Update to Asterisk 11.17.1: this contains a security fix, plus various bugs. pkgsrc changes: - adapt to upstream support for clang - more comprehensive sweep for 64-bit time_t related stuff - XXX pjsip has its own time related stuff that is 32-bit only ----- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. The release of these versions resolves the following security vulnerability: * AST-2015-003: TLS Certificate Common name NULL byte exploit When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack. For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 The security advisory is available at: * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.17.0. The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE with replaces (Reported by Eelco Brolman) * 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-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-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-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * 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-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-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-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-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-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-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-11.17.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.16.0. The release of Asterisk 11.16.0 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-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * 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-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-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-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-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-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-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * 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-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0 Thank you for your continued support of Asterisk!
2015-05-19 09:52:14 +02:00
sscanf(remainder, "%79s", timezone);
tv = ast_mktime(&tm, S_OR(timezone, NULL));
- snprintf(result, sizeof(result), "%ld", (long) tv.tv_sec);
+ snprintf(result, sizeof(result), "%jd", (intmax_t) tv.tv_sec);
pbx_builtin_setvar_helper(chan, "__TESTTIME", result);
return 0;
}
Update Asterisk to 11.20.0: this is mainly a bug fix release. pkgsrc changes: - from joerg@ - srtp support - new asterisk-config option to control installing of sample config files - manifest.xml for Solaris' SMF - various bugfixes, some reworked by myself - backport kqueue timer update from Asterisk 13 ----- The Asterisk Development Team has announced the release of Asterisk 11.20.0. The release of Asterisk 11.20.0 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-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-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * 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-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk) * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp) * ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz) * 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-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström) Improvements made in this release: ----------------------------------- * 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-11.20.0 Thank you for your continued support of Asterisk!
2015-10-27 09:49:01 +01:00
@@ -11446,7 +11446,7 @@ static int pbx_builtin_gotoiftime(struct
struct ast_timing timing;
const char *ctime;
struct timeval tv = ast_tvnow();
- long timesecs;
+ intmax_t timesecs;
if (!chan) {
ast_log(LOG_WARNING, "GotoIfTime requires a channel on which to operate\n");
@@ -11461,7 +11461,7 @@ static int pbx_builtin_gotoiftime(struct
Update to Asterisk 11.17.1: this contains a security fix, plus various bugs. pkgsrc changes: - adapt to upstream support for clang - more comprehensive sweep for 64-bit time_t related stuff - XXX pjsip has its own time related stuff that is 32-bit only ----- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. The release of these versions resolves the following security vulnerability: * AST-2015-003: TLS Certificate Common name NULL byte exploit When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack. For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 The security advisory is available at: * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.17.0. The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE with replaces (Reported by Eelco Brolman) * 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-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-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-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * 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-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-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-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-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-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-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-11.17.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 11.16.0. The release of Asterisk 11.16.0 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-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * 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-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-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-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-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-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-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * 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-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0 Thank you for your continued support of Asterisk!
2015-05-19 09:52:14 +02:00
ts = s = ast_strdupa(data);
ast_channel_lock(chan);
- if ((ctime = pbx_builtin_getvar_helper(chan, "TESTTIME")) && sscanf(ctime, "%ld", &timesecs) == 1) {
+ if ((ctime = pbx_builtin_getvar_helper(chan, "TESTTIME")) && sscanf(ctime, "%jd", &timesecs) == 1) {
tv.tv_sec = timesecs;
} else if (ctime) {
ast_log(LOG_WARNING, "Using current time to evaluate\n");
Reference in a new issue Copy permalink