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@@ -1,3 +1,809 @@
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2015-01-28 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert10 Released.
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* Mitigate possible HTTP injection attacks using CURL() function in
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Asterisk.
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CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request
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injection can be performed given properly-crafted URLs.
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Since Asterisk makes use of libcURL, and it is possible that users of
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Asterisk may get cURL URLs from user input or remote sources, we have
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made a patch to Asterisk to prevent such HTTP injection attacks from
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originating from Asterisk.
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ASTERISK-24676 #close
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Reported by: Matt Jordan, Olle Johansson
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Review: https://reviewboard.asterisk.org/r/4364
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AST-2015-002
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2014-12-10 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert9 Released.
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* AST-2014-019: Fix crash when receiving a WebSocket packet with a
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payload length of zero.
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Frames with a payload length of 0 were incorrectly handled in
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res_http_websocket. Provided a frame with a payload had been
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received prior it was possible for a double free to occur. The
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realloc operation would succeed (thus freeing the payload) but be
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treated as an error. When the session was then torn down the payload
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would be freed again causing a crash. The read function now takes
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this into account.
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This change also fixes assumptions made by users of
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res_http_websocket. There is no guarantee that a frame received from
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it will be NULL terminated.
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ASTERISK-24472 #close
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Reported by: Badalian Vyacheslav
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2014-11-20 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert8 Released.
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* AST-2014-012: Fix error with mixed address family ACLs.
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Prior to this commit, the address family of the first item in an ACL
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was used to compare all incoming traffic. This could lead to traffic
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of other IP address families bypassing ACLs.
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ASTERISK-24469 #close
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Reported by Matt Jordan
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* AST-2014-014: Fix race condition where channels may get stuck in
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ConfBridge under load.
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Under load it was possible for the bridging API, and thus ConfBridge,
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to get channels that may have hung up stuck in it. This is because
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handling of state transitions for a bridged channel within a bridge
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was not protected and simply set the new state without regard to the
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existing state. If the existing state had been hung up this would get
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overwritten.
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This change adds locking to protect changing of the state and also
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takes into consideration the existing state.
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ASTERISK-24440 #close
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Reported by: Ben Klang
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* AST-2014-017 - app_confbridge: permission escalation/ class
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authorization.
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Confbridge dialplan function permission escalation via AMI and
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inappropriate class authorization on the ConfbridgeStartRecord action.
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The CONFBRIDGE dialplan function when executed from an external
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protocol (for instance AMI), could result in a privilege escalation.
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Also, the AMI action “ConfbridgeStartRecord” could also be used to
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execute arbitrary system commands without first checking for system
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access.
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Asterisk now inhibits the CONFBRIDGE function from being executed
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from an external interface if the live_dangerously option is set to
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no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed
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to execute under a user with system level access.
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ASTERISK-24490
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Reported by: Gareth Palmer
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* AST-2014-018 - func_db: DB Dialplan function permission escalation
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via AMI.
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The DB dialplan function when executed from an external protocol
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(for instance AMI), could result in a privilege escalation.
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Asterisk now inhibits the DB function from being executed from an
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external interface if the live_dangerously option is set to no.
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ASTERISK-24534
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Reported by: Gareth Palmer
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patches: submitted by Gareth Palmer (license 5169)
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2014-10-20 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert7 Released.
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* AST-2014-011: Fix POODLE security issues
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There are two aspects to the vulnerability:
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(1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module
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to use TLSv1+. At this time, it does not refactor res_jabber/
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res_xmpp to use the TCP/TLS core, which should be done as an
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improvement at a latter date.
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(2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left
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unspecified, will default to the OpenSSL SSLv23_method. This
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method allows for all encryption methods, including SSLv2/SSLv3.
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A MITM can exploit this by forcing a fallback to SSLv3, which
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leaves the server vulnerable to POODLE. This patch adds WARNINGS
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if a user uses SSLv2/SSLv3 in their configuration, and explicitly
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disables SSLv2/SSLv3 if using SSLv23_method.
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For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or
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SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer
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support SSLv2 or SSLv3.
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Much thanks to abelbeck for reporting the vulnerability and providing
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a patch for the res_jabber/res_xmpp modules.
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2014-09-18 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert6 Released.
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* AST-2014-010: Resolve crash when the Message channel technology
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enters into the ReceiveFax application using res_fax_spandsp
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If faxing fails at a very early stage, then it is possible for
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us to pass a NULL t30 state pointer to spandsp, which spandsp
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is none too pleased with.
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This patch ensures that we pass the correct pointer to spandsp
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in the situation where we have not yet set our local t30 state
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pointer.
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An advisory was made for this issue due to the likelihood of
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it occurring in some Asterisk configurations.
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ASTERISK-24301 #close
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Reported by Matt Jordan, Philippe Lindheimer
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2014-09-05 Asterisk Development Team <asteriskteam@digium.com>
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* Certified Asterisk 11.6-cert5 Released.
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2014-08-17 01:54 +0000 [r421209] Kinsey Moore <kmoore@digium.com>
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* res/res_snmp.c, apps/app_dictate.c, apps/app_test.c,
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apps/app_ices.c, res/res_http_websocket.c, cdr/cdr_radius.c,
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build_tools/cflags.xml, funcs/func_pitchshift.c,
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apps/app_osplookup.c, funcs/func_frame_trace.c,
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channels/console_gui.c, apps/app_mp3.c, pbx/pbx_ael.c,
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channels/console_board.c, formats/format_jpeg.c,
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channels/chan_mgcp.c, res/res_config_pgsql.c, cel/cel_tds.c,
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apps/app_dahdiras.c, res/res_ael_share.c, apps/app_talkdetect.c,
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utils/conf2ael.c, channels/chan_jingle.c, channels/chan_misdn.c,
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formats/format_vox.c, res/res_timing_pthread.c,
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res/res_corosync.c, cel/cel_sqlite3_custom.c, apps/app_sms.c,
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apps/app_zapateller.c, res/res_fax_spandsp.c,
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res/res_timing_kqueue.c, utils/check_expr.c,
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channels/chan_unistim.c, build_tools/cflags-devmode.xml,
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utils/muted.c, cdr/cdr_sqlite3_custom.c, res/res_phoneprov.c,
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channels/console_video.c, apps/app_alarmreceiver.c,
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apps/app_chanisavail.c, apps/app_image.c, channels/chan_gtalk.c,
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cdr/cdr_pgsql.c, res/res_config_sqlite.c, res/res_pktccops.c,
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cdr/cdr_csv.c, utils/stereorize.c, channels/chan_phone.c,
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channels/chan_skinny.c, build_tools/embed_modules.xml,
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apps/app_minivm.c, pbx/pbx_realtime.c, apps/app_amd.c,
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channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
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cdr/cdr_odbc.c, res/res_config_ldap.c, apps/app_jack.c,
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apps/app_adsiprog.c, utils/refcounter.c, apps/app_nbscat.c,
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apps/app_festival.c, apps/app_waitforsilence.c, utils/astman.c,
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apps/app_morsecode.c, utils/smsq.c, pbx/pbx_lua.c,
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channels/chan_console.c, apps/app_getcpeid.c,
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channels/chan_oss.c, cdr/cdr_tds.c, apps/app_waitforring.c,
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pbx/pbx_dundi.c, utils/ael_main.c, utils/extconf.c,
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channels/chan_nbs.c, utils/streamplayer.c, cel/cel_pgsql.c,
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cel/cel_radius.c: Add missing commit from 11.2-cert This disables
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building by default for all extended modules for Certified
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Asterisk 11.6. This commit was missed from 11.2-cert when
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creating the 11.6-cert branch. ASTERISK-24104 #close Reported by:
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Rusty Newton
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2014-08-08 17:18 +0000 [r420559] Richard Mudgett <rmudgett@digium.com>
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* /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
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resolve the large SDP poll issue. Replace sip_tls_read() and
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sip_tcp_read() with a single function and resolve the poll/wait
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issue with large SDP payloads. ASTERISK-18345 #close Reported by:
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Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
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patch uploaded by Elazar Broad Review:
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https://reviewboard.asterisk.org/r/3882/ ........ Merged
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revisions 420434 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 420435 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-07-25 23:27 +0000 [r419662] Richard Mudgett <rmudgett@digium.com>
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* main/features.c, /: features.c: Allow appliationmap to use Gosub.
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Using DYNAMIC_FEATURES with a Gosub application as the mapped
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application does not work. It does not work because Gosub just
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pushes the current dialplan context, exten, and priority onto a
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stack and sets the specified Gosub location. Gosub does not have
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a dialplan execution loop to run dialplan like Macro. * Made the
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DYNAMIC_FEATURES application mapping feature call
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ast_app_exec_macro() and ast_app_exec_sub() for the Macro and
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Gosub applications respectively. * Backported
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ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute
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dialplan routines from the DYNAMIC_FEATURES application mapping
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feature. NOTE: This issue does not affect v12+ because it already
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does what this patch implements. AST-1391 #close Reported by:
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Guenther Kelleter Review:
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https://reviewboard.asterisk.org/r/3844/ ........ Merged
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revisions 419630 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 419631 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-07-23 14:34 +0000 [r419308] Scott Griepentrog <sgriepentrog@digium.com>
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* /, apps/app_voicemail.c: app_voicemail: use a consistent
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generator string When updating voicemail.conf when a user changes
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their pin, change the generator string to be the same as the
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module name when reading so that the same config_hook will be
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called. Review: https://reviewboard.asterisk.org/r/3837/ ........
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Merged revisions 419284 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-07-11 16:39 +0000 [r418368] Scott Griepentrog <sgriepentrog@digium.com>
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* /, main/config.c: config: inform config hook of change when
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writing file When updated configuration is written back to the
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conf file - for example when a user changes their voicemail pin,
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make sure that any config hook that wants to know of changes is
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informed. Review: https://reviewboard.asterisk.org/r/3708/
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........ Merged revisions 418366 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-07-01 15:37 +0000 [r417724] Joshua Colp <jcolp@digium.com>
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* res/res_rtp_asterisk.c, main/rtp_engine.c, /,
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channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample,
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include/asterisk/rtp_engine.h, channels/sip/include/sip.h:
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Multiple revisions
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402345,405234,409129-409130,409565,413008,417141,417677 ........
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r402345 | kmoore | 2013-11-01 05:31:49 -0700 (Fri, 01 Nov 2013) |
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11 lines chan_sip: Fix RTCP port for SRFLX ICE candidates This
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corrects one-way audio between Asterisk and Chrome/jssip as a
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result of Asterisk inserting the incorrect RTCP port into RTCP
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SRFLX ICE candidates. This also exposes an ICE component
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enumeration to extract further details from candidates. (closes
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issue ASTERISK-21383) Reported by: Shaun Clark Review:
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https://reviewboard.asterisk.org/r/2967/ ........ r405234 |
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kharwell | 2014-01-09 08:49:55 -0800 (Thu, 09 Jan 2014) | 19
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lines res_rtp_asterisk: Fails to resume WebRTC call from hold In
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ast_rtp_ice_start if the ice session create check list failed,
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start check was never initiated and ice_started was never set to
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true. Upon re-entering the function (for instance, [un]hold) it
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would try to create the check list again with duplicate remote
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candidates. Fixed so that if the create check list fails the
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necessary data structures are properly re-initialized for any
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subsequent retries. Note, it was decided to not stop ice support
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(by calling ast_rtp_ice_stop) on a check list failure because it
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possible things might still work. However, a debug message was
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added to help with any future troubleshooting. (closes issue
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ASTERISK-22911) Reported by: Vytis Valentinavičius Patches:
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works_on_my_machine.patch uploaded by xytis (license 6558)
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........ r409129 | jrose | 2014-02-27 11:19:02 -0800 (Thu, 27 Feb
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2014) | 15 lines res_rtp_asterisk: Fix checklist creating
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problems in ICE sessions Prior to this patch, local candidate
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lists including SRFLX would fail to start properly when building
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ICE candidate check lists. This patch fixes that problem by
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making sure that each SRFLX candidate is associated with the
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proper base address so that the check list can create matches
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properly. This patch was written by jcolp. The issue will be left
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open to await testing by the issue participants. (issue
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ASTERISK-23213) Reported by: Andrea Suisani Review:
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https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose
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| 2014-02-27 11:38:10 -0800 (Thu, 27 Feb 2014) | 8 lines
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res_rtp_asterisk: correct build error from r409129 Accidentally
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placed a declaration below functional code (issue ASTERISK-23213)
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Reported by: Andrea Suisani Review:
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https://reviewboard.asterisk.org/r/3256/ ........ r409565 | jrose
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| 2014-03-04 08:40:39 -0800 (Tue, 04 Mar 2014) | 9 lines
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res_rtp_asterisk: Fix one way audio problems with hold/unhold
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when using ICE ICE sessions will now be restarted if sessions are
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changed to use new sets of remote candidates. (closes issue
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ASTERISK-22911) Reported by: Vytis Valentinavičius Review:
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https://reviewboard.asterisk.org/r/3275/ ........ r413008 |
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mjordan | 2014-04-25 10:47:21 -0700 (Fri, 25 Apr 2014) | 14 lines
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res_rtp_asterisk: Add support for DTLS handshake retransmissions
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On congested networks, it is possible for the DTLS handshake
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messages to get lost. This patch adds a timer to res_rtp_asterisk
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that will periodically check to see if the handshake has
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succeeded. If not, it will retransmit the DTLS handshake. Review:
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https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close
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Reported by: Nitesh Bansal patches: dtls_retransmission.patch
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uploaded by Nitesh Bansal (License 6418) ........ r417141 | file
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| 2014-06-23 11:49:14 -0700 (Mon, 23 Jun 2014) | 5 lines
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res_rtp_asterisk: Return the length of data written when sending
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via ICE instead of 0. ASTERISK-23834 #close Reported by: Richard
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Kenner ........ r417677 | file | 2014-06-30 12:42:18 -0700 (Mon,
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30 Jun 2014) | 12 lines res_rtp_asterisk: Add SHA-256 support for
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DTLS and perform DTLS negotiation on RTCP. This change fixes up
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DTLS support in res_rtp_asterisk so it can accept and provide a
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SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after
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ICE negotiation completes. Configuration options to chan_sip have
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also been added to allow behavior to be tweaked (such as forcing
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the AVP type media transports in SDP). ASTERISK-22961 #close
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Reported by: Jay Jideliov Review:
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https://reviewboard.asterisk.org/r/3679/ ........ Merged
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revisions 402345,405234,409129-409130,409565,413008,417141,417677
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from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-06-13 05:29 +0000 [r415977-416106] Richard Mudgett <rmudgett@digium.com>
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* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
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main/http.c, include/asterisk/tcptls.h: AST-2014-007: Fix of fix
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to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close
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Reported by: Richard Mudgett Review:
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https://reviewboard.asterisk.org/r/3617/ ........ Merged
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revisions 416066 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 416067 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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* main/http.c, UPGRADE.txt, main/utils.c,
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include/asterisk/tcptls.h, res/res_http_websocket.c,
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configs/http.conf.sample, include/asterisk/utils.h,
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main/tcptls.c, main/manager.c, /, channels/chan_sip.c:
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AST-2014-007: Fix DOS by consuming the number of allowed HTTP
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connections. Simply establishing a TCP connection and never
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sending anything to the configured HTTP port in http.conf will
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tie up a HTTP connection. Since there is a maximum number of open
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HTTP sessions allowed at a time you can block legitimate
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connections. A similar problem exists if a HTTP request is
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started but never finished. * Added http.conf session_inactivity
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timer option to close HTTP connections that aren't doing
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anything. Defaults to 30000 ms. * Removed the undocumented
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manager.conf block-sockets option. It interferes with TCP/TLS
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inactivity timeouts. * AMI and SIP TLS connections now have
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better authentication timeout protection. Though I didn't remove
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the bizzare TLS timeout polling code from chan_sip. * chan_sip
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can now handle SSL certificate renegotiations in the middle of a
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session. It couldn't do that before because the socket was
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non-blocking and the SSL calls were not restarted as documented
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by the OpenSSL documentation. * Fixed an off nominal leak of the
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ssl struct in handle_tcptls_connection() if the FILE stream
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failed to open and the SSL certificate negotiations failed. The
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patch creates a custom FILE stream handler to give the created
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FILE streams inactivity timeout and timeout after a specific
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moment in time capability. This approach eliminates the need for
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code using the FILE stream to be redesigned to deal with the
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timeouts. This patch indirectly fixes most of ASTERISK-18345 by
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fixing the usage of the SSL_read/SSL_write operations.
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ASTERISK-23673 #close Reported by: Richard Mudgett ........
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Merged revisions 415841 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 415854 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-06-12 16:27 +0000 [r415867] Scott Griepentrog <sgriepentrog@digium.com>
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* /, apps/app_queue.c: app_queue: delayed state can cause early
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leavewhenempty ringing In app_queue, device state changes arrive
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in event messages and update the queue member status value. That
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value is checked in get_member_status() to decide that the caller
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should leave when there are no available members. Although event
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messages can be delayed by other activity, there is no adverse
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affect by lagged status except in one specific case: there is
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only one available member, it was just rung, and leavewhenempty
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is enabled set for ringing members. This change adds a direct
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check of the device state only under this condition where the
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caller may be dropped incorrectly, resolving this issue without
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affecting performance of app_queue normally. AST-1248 #close
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Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
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Thomas Arimont ........ Merged revisions 415833 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8
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2014-06-12 16:06 +0000 [r415842] Jonathan Rose <jrose@digium.com>
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* /, UPGRADE.txt, apps/app_mixmonitor.c: MixMonitor: Add class
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authorization requirements to MixMonitor AMI commands MixMonitor
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AMI commands StartMixMonitor and StopMixMonitor lacked class
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authorization. StopMixMonitor now requires that the manager user
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either have the call or system class authorization.
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StartMixMonitor is a slightly larger issue since it can execute
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shell commands if the right arguments are passed into it, and we
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consider this a permission escalation. A security release will be
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issued for problem this shortly. ASTERISK-23609 #close Reported
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by: Corey Farrell ........ Merged revisions 415837 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-05-13 00:48 +0000 [r413773] Richard Mudgett <rmudgett@digium.com>
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* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
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channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
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PROGRESS events when overlap dialing is enabled. When overlap
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dialing is enabled, the lack of inband audio available
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information in the SETUP_ACKNOWLEDGE events causes an
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interoperability problem with SIP. sig_pri doesn't know if there
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is dialtone present when a SETUP_ACKNOWLEDGE is received so it
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assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
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SIP channel driver then sends out a 183 Session Progress and
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blocks the desired 180 Ringing message when the ALERTING message
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comes in. * Made the configure script detect if the installed
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|
version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
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Using the new API, made generate an AST_CONTROL_PROGRESS frame on
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an incoming SETUP_ACKNOWLEDGE message when the message indicates
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inband audio is present instead of assuming that dialtone is
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present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
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inband audio available indication only if dialtone is expected.
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|
The change also makes the fallback behaviour of sending the
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PROGRESS message better by sending it only if dialtone is
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expected. * Changed receiving a PROCEEDING message to not
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|
generate an AST_CONTROL_PROGRESS frame if the progress indication
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ie indicates non-end-to-end-ISDN. This helps interoperability
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with SIP. * Changed sending a PROCEEDING message in response to
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an AST_CONTROL_PROCEEDING frame to not indicate inband audio
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|
available. It was silly to do so anyway because the channel
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|
|
driver doesn't know if inband audio is even available. This helps
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|
interoperability with SIP. This patch and a corresponding change
|
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|
in libpri work together to allow Asterisk to control the inband
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audio available progress indication ie on the SETUP_ACKNOWLEDGE
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message when dialtone is present. AST-1338 #close Reported by:
|
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|
|
Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
|
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|
|
|
........ Merged revisions 413714 from
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|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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|
revisions 413765 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-04-11 17:27 +0000 [r412212] Kevin Harwell <kharwell@digium.com>
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|
* main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
|
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|
on rasterisk Even since the fixes of AST-2013-007, Asterisk
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|
prints the following warning on startup if the user decided to
|
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|
|
live dangerously: Privilege escalation protection disabled! See
|
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|
|
https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
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|
message is intended for the logs and interactive startup. No need
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|
|
for it to appear on a remote console. This commit removes it from
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there. (closes issue ASTERISK-23084) Review:
|
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|
https://reviewboard.asterisk.org/r/3101/ ........ Merged
|
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|
revisions 404861 from
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|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
|
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|
revisions 404888 from
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http://svn.asterisk.org/svn/asterisk/branches/11
|
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|
2014-03-10 17:34 +0000 [r410429] Richard Mudgett <rmudgett@digium.com>
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|
* /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
|
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|
|
of Cookie headers. Sending a HTTP request that is handled by
|
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|
|
Asterisk with a large number of Cookie headers could overflow the
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|
stack. Another vulnerability along similar lines is any HTTP
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request with a ridiculous number of headers in the request could
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exhaust system memory. (closes issue ASTERISK-23340) Reported by:
|
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|
|
Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
|
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|
|
|
Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
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|
410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
|
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|
........ Merged revisions 410381 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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|
2014-03-10 14:04 +0000 [r410359] Kinsey Moore <kmoore@digium.com>
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* /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
|
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|
|
session timers request This change allows chan_sip to avoid
|
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creation of the channel and consumption of associated file
|
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descriptors altogether if the inbound request is going to be
|
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|
rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
|
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|
|
Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
|
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|
|
Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
|
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|
|
Corey Farrell (license 5909) ........ Merged revisions 410308
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|
|
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
|
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|
|
Merged revisions 410311 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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|
2014-02-19 19:17 +0000 [r408392] Richard Mudgett <rmudgett@digium.com>
|
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|
* main/config.c, /: config: Add file size and nanosecond resolution
|
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|
fields to the cached modified config file information. Repeatedly
|
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|
modifying config files and reloading too fast sometimes fails to
|
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|
reload the configuration because the cached modification
|
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|
timestamp has one second resolution. * Added file size and
|
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nanosecond resolution fields to the cached config file
|
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|
modification timestamp information. Now if the file size changes
|
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|
or the file system supports nanosecond resolution the modified
|
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|
|
file has a better chance of being detected for reload. * Added a
|
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|
|
missing unlock in an off-nominal code path. (closes issue
|
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|
|
AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
|
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|
|
|
........ Merged revisions 408387 from
|
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|
|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
|
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|
|
revisions 408388 from
|
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|
http://svn.asterisk.org/svn/asterisk/branches/11
|
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|
2014-02-07 19:30 +0000 [r407746] Richard Mudgett <rmudgett@digium.com>
|
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|
|
* channels/chan_iax2.c, include/asterisk/frame.h,
|
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|
|
configs/iax.conf.sample, /: chan_iax2: Block unnecessary control
|
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|
|
frames to/from the wire. Establishing an IAX2 call between
|
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|
|
Asterisk v1.4 and v1.8 (or later) results in an unexpected call
|
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|
disconnect. The problem happens because newer values in the enum
|
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|
|
ast_control_frame_type are not consistent between the branch
|
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|
|
versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
|
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|
|
using IAX2 2) v1.8 answers and sends a connected line update
|
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|
control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
|
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|
|
receives the control frame as an end-of-q (on v1.4
|
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|
AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
|
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|
|
receive queue becomes empty. Several things are done by this
|
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|
|
|
patch to fix the problem and attempt to prevent it from happening
|
|
|
|
|
again in the future: * Added a warning at the definition of enum
|
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|
|
ast_control_frame_type about how to add new control frame values.
|
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|
* Made block sending and receiving control frames that have no
|
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|
reason to go over the wire. * Extended the connectedline iax.conf
|
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|
parameter to also include the redirecting information updates. *
|
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|
Updated the connectedline iax.conf parameter documentation to
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|
include a notice that the parameter must be "no" when the peer is
|
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|
|
an Asterisk v1.4 instance. (closes issue AST-1302) Review:
|
|
|
|
|
https://reviewboard.asterisk.org/r/3174/ ........ Merged
|
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|
|
revisions 407678 from
|
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|
|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
|
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|
revisions 407727 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-01-25 00:13 +0000 [r406358-406469] Richard Mudgett <rmudgett@digium.com>
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* /, main/cel.c: CEL: Protect data structures during reload and
|
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shutdown. The CEL data structures need to be protected during a
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configuration reload and shutdown. Asterisk crashed during a
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shutdown because CEL events were still in flight and the CEL data
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structures were already destroyed. * Protected the appset and
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linkedids ao2 containers using the reload_lock. As a result
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appset, linkedids, and held objects don't need a lock. * Added
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NULL checks before use of the appset and linkedids ao2 containers
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in case the CEL module is already shutdown. * Fixed overloading
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of the linkedids held objects reference count. During shutdown
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any held objects would be leaked. * Fixed memory leak of
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linkedids held objects if the LINKEDID_END is not being tracked.
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The objects in the linkedids container were not removed if the
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LINKEDID_END event is not used. * Added access protection to the
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appset container during the CLI "cel show status" command. * Made
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CEL config reload not set defaults if the cel.conf file is
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invalid. (closes issue AST-1253) Reported by: Guenther Kelleter
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Review: https://reviewboard.asterisk.org/r/3127/ ........ Merged
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revisions 406417 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 406418 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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* main/manager.c, /: manager: Protect data structures during
|
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shutdown. Occasionally, the manager module would get an
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"INTERNAL_OBJ: bad magic number" error on a "core restart
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gracefully" command if an AMI connection is established. * Added
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ao2_global_obj protection to the sessions global container. *
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Fixed the order of unreferencing a session object in
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session_destroy(). * Removed unnecessary container traversals of
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the white/black filters during session_destructor(). (closes
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issue AST-1242) Reported by: Guenther Kelleter Review:
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https://reviewboard.asterisk.org/r/3144/ ........ Merged
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revisions 406341 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-01-15 15:27 +0000 [r405536-405578] Matthew Jordan <mjordan@digium.com>
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* main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
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prevent memory corruption During dialplan execution in
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pbx_extension_helper(), the contexts global read lock prevents
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link list corruption, but was released with a pointer to the
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ast_exten and data later used in variable substitution. Instead,
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this patch removes pbx_substitute_variables() and locates a copy
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of the ast_exten data on the stack before releasing the lock,
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where ast_exten could get free'd by another thread performing a
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module reload. (issue AST-1179) Reported by: Thomas Arimont
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(issue AST-1246) Reported by: Alexander Hömig Review:
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https://reviewboard.asterisk.org/r/3055/ ........ Merged
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revisions 403862 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 403863 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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* /, channels/chan_sip.c: chan_sip: Hangup transferer/transferee
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when transfer to Parking fails When performing a SIP transfer to
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a Park extension, if the Park fails, chan_sip will currently not
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hang up either the transferer or the transfer target. This
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results in the channels being orphaned with no thread to service
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frames, resulting in stuck channels. This patch immediately hangs
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up the two channels if a Park fails. (closes issue
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ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes
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issue ASTERISK-23047) Reported by: Tommy Thompson Tested by:
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Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107
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........ Merged revisions 405380 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-01-14 18:50 +0000 [r405488] Richard Mudgett <rmudgett@digium.com>
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* apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample,
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main/cli.c, include/asterisk/logger.h, main/pbx.c,
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main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c,
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main/logger.c, UPGRADE.txt: verbosity: Fix performance of console
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verbose messages. The per console verbose level feature as
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previously implemented caused a large performance penalty. The
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fix required some minor incompatibilities if the new rasterisk is
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used to connect to an earlier version. If the new rasterisk
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connects to an older Asterisk version then the root console
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verbose level is always affected by the "core set verbose"
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command of the remote console even though it may appear to only
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affect the current console. If an older version of rasterisk
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connects to the new version then the "core set verbose" command
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will have no effect. * Fixed the verbose performance by not
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generating a verbose message if nothing is going to use it and
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then filtered any generated verbose messages before actually
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sending them to the remote consoles. * Split the "core set debug"
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and "core set verbose" CLI commands to remove the per module
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verbose support that cannot work with the per console verbose
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level. * Added a silent option to the "core set verbose" command.
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* Fixed "core set debug off" tab completion. * Made "core show
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settings" list the current console verbosity in addition to the
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root console verbosity. * Changed the default verbose level of
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the 'verbose' setting in the logger.conf [logfiles] section. The
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default is now to once again follow the current root console
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level. As a result, using the AMI Command action with "core set
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verbose" could again set the root console verbose level and
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affect the verbose level logged. (closes issue AST-1252) Reported
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by: Guenther Kelleter Review:
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|
https://reviewboard.asterisk.org/r/3114/ ........ Merged
|
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|
|
revisions 405431 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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2014-01-09 16:34 +0000 [r405233] Matthew Jordan <mjordan@digium.com>
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* /, apps/app_confbridge.c,
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apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
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|
crash caused when waitmarked/marked users leave together When
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waitmarked users join a ConfBridge, the conference state is
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transitioned from EMPTY -> INACTIVE. In this state, the users are
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maintined in a waiting users list. When a marked user joins, the
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ConfBridge conference transitions from INACTIVE -> MULTI_MARKED,
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and all users are put onto the active list of users. This process
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works correctly. When the marked user leaves, if they are the
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last marked user, the MULTI_MARKED state does the following: (1)
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|
It plays back a message to the bridge stating that the leader has
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|
left the conference. This requires an unlocking of the bridge.
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(2) It moves waitmarked users back to the waiting list (3) It
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|
transitions to the appropriate state: in this case, INACTIVE
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|
However, because it plays the prompt back to the bridge before
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moving the users and before finishing the state transition, this
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|
|
creates a race condition: with the bridge unlocked, waitmarked
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|
users who leave the conference (or are kicked from it) can cause
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|
|
a state transition of the bridge to another state before the
|
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|
|
conference is transitioned to the INACTIVE state. This causes the
|
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|
|
state machine to get a bit wonky, often leading to a crash when
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|
the MULTI_MARKED state attempts to conclude its processing. This
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patch fixes this problem: (1) It prevents kicked users from being
|
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|
kicked again. That's just a nicety. (2) More importantly, it
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|
|
fixes the race condition by only playing the prompt once the
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|
state has transitioned correctly to INACTIVE. If waitmarked users
|
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|
|
sneak out during the prompt being played, no harm no foul.
|
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|
Review: https://reviewboard.asterisk.org/r/3108/ (closes issue
|
|
|
|
|
AST-1258) Reported by: Steve Pitts ........ Merged revisions
|
|
|
|
|
405215 from http://svn.asterisk.org/svn/asterisk/branches/11
|
|
|
|
|
|
|
|
|
|
2013-12-19 16:38 +0000 [r404349] Scott Griepentrog <sgriepentrog@digium.com>
|
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|
|
|
|
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|
|
* main/db.c, /: astdb: crash in sqlite3 during shutdown When
|
|
|
|
|
Asterisk is shut down, the astdb_atexit() function releases
|
|
|
|
|
(finalize) the previously initiated (prepared) SQL statements in
|
|
|
|
|
sqlite3. Another thread making a subsequent request can cause a
|
|
|
|
|
crash in sqlite3. This patch eliminates that issue by resetting
|
|
|
|
|
the statement pointer after it is released/cleared. The sqlite3
|
|
|
|
|
code detects the null pointer, and aborts the operation cleanly.
|
|
|
|
|
(closes issue AST-1265) Reported by: Alexander Hömig (closes
|
|
|
|
|
issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
|
|
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|
|
Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged
|
|
|
|
|
revisions 404344 from
|
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|
|
http://svn.asterisk.org/svn/asterisk/branches/11
|
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|
|
|
|
|
|
|
|
2013-12-16 17:29 +0000 [r403956] David M. Lee <dlee@digium.com>
|
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|
|
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|
|
* funcs/func_realtime.c, main/pbx.c, main/tcptls.c,
|
|
|
|
|
funcs/func_db.c, /, README-SERIOUSLY.bestpractices.txt,
|
|
|
|
|
configs/asterisk.conf.sample, funcs/func_shell.c,
|
|
|
|
|
funcs/func_env.c, funcs/func_lock.c, UPGRADE.txt,
|
|
|
|
|
include/asterisk/pbx.h, main/asterisk.c: security: Inhibit
|
|
|
|
|
execution of privilege escalating functions This patch allows
|
|
|
|
|
individual dialplan functions to be marked as 'dangerous', to
|
|
|
|
|
inhibit their execution from external sources. A 'dangerous'
|
|
|
|
|
function is one which results in a privilege escalation. For
|
|
|
|
|
example, if one were to read the channel variable SHELL(rm -rf /)
|
|
|
|
|
Bad Things(TM) could happen; even if the external source has only
|
|
|
|
|
read permissions. Execution from external sources may be enabled
|
|
|
|
|
by setting 'live_dangerously' to 'yes' in the [options] section
|
|
|
|
|
of asterisk.conf. Although doing so is not recommended. (closes
|
|
|
|
|
issue ASTERISK-22905) Review:
|
|
|
|
|
http://reviewboard.digium.internal/r/432/ ........ Merged
|
|
|
|
|
revisions 403913 from
|
|
|
|
|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
|
|
|
|
|
revisions 403917 from
|
|
|
|
|
http://svn.asterisk.org/svn/asterisk/branches/11
|
|
|
|
|
|
|
|
|
|
2013-12-16 15:38 +0000 [r403860] Scott Griepentrog <sgriepentrog@digium.com>
|
|
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|
|
|
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|
|
* apps/app_sms.c: app_sms: BufferOverflow when receiving odd length
|
|
|
|
|
16 bit message This patch prevents an infinite loop overwriting
|
|
|
|
|
memory when a message is received into the unpacksms16()
|
|
|
|
|
function, where the length of the message is an odd number of
|
|
|
|
|
bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens
|
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|
|
Tested by: Jan Juergens
|
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|
|
2013-11-04 21:20 +0000 [r402463] Kevin Harwell <kharwell@digium.com>
|
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|
* /, channels/chan_sip.c: chan_sip: notify dialog info ignores
|
|
|
|
|
presentation indicator in callerid The presentation indicator in
|
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|
|
|
a callerid (e.g. set by dialplan function
|
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|
|
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
|
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|
|
|
Info Notifies are generated during extension monitoring. Added a
|
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|
|
check to make sure the name and/or number presentations on the
|
|
|
|
|
callee (remote identity) are set to allow. If they are restricted
|
|
|
|
|
then "anonymous" is used instead. (closes issue AST-1175)
|
|
|
|
|
Reported by: Thomas Arimont Review:
|
|
|
|
|
https://reviewboard.asterisk.org/r/2976/ ........ Merged
|
|
|
|
|
revisions 402450 from
|
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|
|
http://svn.asterisk.org/svn/asterisk/branches/11
|
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|
|
2013-11-01 20:39 +0000 [r402377-402383] Matthew Jordan <mjordan@digium.com>
|
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|
|
* asterisk-11.6.0-summary.html (removed),
|
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|
|
|
asterisk-11.6.0-summary.txt (removed): Remove old summaries
|
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|
|
* include/asterisk/pbx.h, res/res_rtp_asterisk.c, main/pbx.c, /,
|
|
|
|
|
configure, configure.ac: Multiple revisions
|
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|
|
|
396884,400075,400093,401446,401960 ........ r396884 | jbigelow |
|
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|
|
2013-08-16 17:45:10 -0500 (Fri, 16 Aug 2013) | 8 lines Add test
|
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|
|
|
suite events to indicate when a feature is detected or not These
|
|
|
|
|
are needed by the bridge test suite tests for them to be able to
|
|
|
|
|
run against Asterisk 11. Review:
|
|
|
|
|
https://reviewboard.asterisk.org/r/2751/ ........ r400075 |
|
|
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|
|
mjordan | 2013-09-28 16:59:12 -0500 (Sat, 28 Sep 2013) | 16 lines
|
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|
|
Add check for openSUSE when detecting bfd library In
|
|
|
|
|
ASTERISK-17842, some additional library checks were added to the
|
|
|
|
|
configure script so that the bfd library could be found on CentOS
|
|
|
|
|
and Fedora systems. As it turns out, openSUSE requires an
|
|
|
|
|
additional library. This patch adds another check to the
|
|
|
|
|
configure script for openSUSE that will add that library. Review:
|
|
|
|
|
https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169)
|
|
|
|
|
Reported by: Guenther Kelleter ........ Merged revisions 400073
|
|
|
|
|
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
|
|
|
|
|
r400093 | mjordan | 2013-09-28 17:21:37 -0500 (Sat, 28 Sep 2013)
|
|
|
|
|
| 23 lines res_rtp_asterisk: Correct erroneous lost packet
|
|
|
|
|
information in RTCP reports RTCP's calculation of the number of
|
|
|
|
|
lost packets in an RTP stream is based on that stream's sequence
|
|
|
|
|
number count, the number of received packets, and how many
|
|
|
|
|
packets we expect to receive. When the SSRC for an RTP stream
|
|
|
|
|
changes, there can - and almost always will be - a large jump in
|
|
|
|
|
the next packet's timestamp and sequence number. If we don't
|
|
|
|
|
reset the number of received packets, sequence number count, and
|
|
|
|
|
other metrics used by RTCP, the next RR/SR report will use the
|
|
|
|
|
previous SSRC's values to calculate the lost packet count for the
|
|
|
|
|
new SSRC - resulting in a very large number of lost packets. This
|
|
|
|
|
patch modifies res_rtp_asterisk such that, if it detects a SSRC
|
|
|
|
|
change, it will reset the various values used by the RTCP
|
|
|
|
|
calculations. From the perspective of RTCP, this appears as a new
|
|
|
|
|
media stream - which is what it is. Review:
|
|
|
|
|
https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
|
|
|
|
|
Reported by: Thomas Arimont ........ Merged revisions 400089 from
|
|
|
|
|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
|
|
|
|
|
r401446 | mjordan | 2013-10-22 17:42:24 -0500 (Tue, 22 Oct 2013)
|
|
|
|
|
| 15 lines res_rtp_asterisk: Fix crash when RTCP is not available
|
|
|
|
|
during SSRC change In r400089, a patch was put in to correct
|
|
|
|
|
erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can
|
|
|
|
|
be called on an RTP instance that does not have RTCP information.
|
|
|
|
|
This patch prevents that crash by only resetting the statistics
|
|
|
|
|
if we do actually have an RTCP instance. (issue AST-1174) (closes
|
|
|
|
|
issue ASTERISK-22667) Reported by: John Bigelow ........ Merged
|
|
|
|
|
revisions 401445 from
|
|
|
|
|
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
|
|
|
|
|
r401960 | sgriepentrog | 2013-10-25 15:44:40 -0500 (Fri, 25 Oct
|
|
|
|
|
2013) | 15 lines pbx.c: fix confused match caller id that deleted
|
|
|
|
|
exten still in hash This fixes a bug where a zero length callerid
|
|
|
|
|
match adjacent to a no match callerid extension entry would be
|
|
|
|
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deleted together, which then resulted in hashtable references to
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free'd memory. A third state of the matchcid value has been added
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to indicate match to any extension which allows enforcing
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comparison of matchcid on/off without errors. (closes issue
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AST-1235) Reported by: Guenther Kelleter Review:
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https://reviewboard.asterisk.org/r/2930/ ........ Merged
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revisions 401959 from
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http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
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revisions 396884,400075,400093,401446,401960 from
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http://svn.asterisk.org/svn/asterisk/branches/11
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* /: SVN properties: Add svnmerge properties for 11
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2013-10-22 16:10 +0000 [r401416] bebuild <bebuild@localhost>:
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* / (added): Create branch for Certified Asterisk 11.6.
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2013-10-21 Asterisk Development Team <asteriskteam@digium.com>
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* Asterisk 11.6.0 Released.
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