23774 Commits (1644cd874ded77f4b9145a39e0224e7046ff39c5)

 

All Branches   

Search

Author	SHA1	Message	Date
Matthew Jordan	1644cd874d	chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line ... This patch is essentially a backport of a small portion of r397526 from ASTERISK-21981. In that patch, pass through support and format attribute negotiation was added for Opus. Part of that included being more tolerant to whitespace in the fmtp line of an SDP; that part of the patch is being applied here. As the author of the backport pointed out, in SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 #close Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 417587 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	23d1b904b0	Ensure REF_DEBUG records entrys for attempts to ao2_ref an invalid object ... This change ensures that __ao2_ref_debug writes to ref_log when given a non-NULL pointer to an invalid ao2 object. This is to ensure that we record any attempt manipulate references of already freed objects. ASTERISK-23948 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3677/ ........ Merged revisions 417500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fe70d27731	refcounter.py: prevent use of excessive RAM with large refs logs ... When processing a 212MB refs file, refcounter.py used over 3GB of RAM. This change greatly reduces memory usage in two ways: * Saving object history in whole lines instead of separated values. * Not saving normal/skewed/leaked object lists unless they are requested. ASTERISK-23921 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3668/ ........ Merged revisions 417480 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5d3078bd48	res_http_websocket: Export symbol for ast_websocket_set_timeout ... Thanks to Sean Bright for pointing out that this was missed in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0b9601052b	udptl: Correct FEC to not consider negative sequence numbers as missing ... When using FEC, with span=3 and entries=4 Asterisk will attempt to repair the packet with sequence number 5, as it will see that packet -4 is missing. The result is Asterisk sending garbage packets that can kill a fax. This patch adds a check to see if the sequence number is valid before checking if the packet is missing. Review: https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close Reported by: Torrey Searle patches: udptl_fec.patch uploaded by Torrey Searle (License 5334) ........ Merged revisions 417318 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	32d684ee25	res_http_websocket: Close websocket correctly and use careful fwrite ... When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2875174e97	chan_sip: Fix handling of "From" headers longer than 256 characters ... From headers were processed using a 256 character buffer on the stack. This change replaces that with a heap allocation by ast_strdup. ASTERISK-23790 #close Reported by: uniken1 Tested by: uniken1 Review: https://reviewboard.asterisk.org/r/3669/ Patches: chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 417248 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3e642e66d1	res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. ... ASTERISK-23834 #close Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	cd6a931be5	main/features - documentation - reformat examples and options in features.conf.sample to show clearly which options apply in which section ... The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. ASTERISK-23667 Review: https://reviewboard.asterisk.org/r/3622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7e9233a81	build: Turn FORTIFY_SOURCE off if DONT_OPTIMIZE is set. ... AST_FORTIFY_SOURCE is automatically set in ./Makefile even if DONT_OPTIMIZE is set in menuselect. This causes gcc to complain that _FORTIFY_SOURCE requires optimization and the build will fail. You can specify "make AST_FORTIFY_SOURCE=''" but I always forget. This patch moves the set of AST_FORTIFY_SOURCE to Makefile.rules and only sets it if DONT_OPTIMIZE is "no". The move is necessary because the top-level Makefile doesn't include menuselect.makeopts. This doesn't solve the entire problem however because res_config_mysql seems to force _FORTIFY_SOURCE so res_config_mysql has to be disabled for now if DONT_OPTIMIZE is set. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3664/ ........ Merged revisions 417016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	2b4423d2b3	build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better. ... ast_ext_tool_check.m4 isn't handling cases where a path to a package is provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config tool (E.G. mysql_config) and the package has its own subdirectories in include or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include. Both cause configure to fail and there are others in the same boat. The problem is caused by logic in ast_ext_tool_check that overrides the result of the config tool's --cflags and --libs options if package_DIR is set. This patch prepends package_DIR (if specified) to the -L and -I results from the package's config tool instead of overriding them. A regenerated ./configure and include/asterisk/autoconfig.h.in are included but can be regenerated by running ./bootstrap.sh at any time. Tested by: George Joseph Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3550/ ........ Merged revisions 416929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a6ba838fce	build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better. ... ast_ext_tool_check.m4 isn't handling cases where a path to a package is provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config tool (E.G. mysql_config) and the package has its own subdirectories in include or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include. Both cause configure to fail and there are others in the same boat. The problem is caused by logic in ast_ext_tool_check that overrides the result of the config tool's --cflags and --libs options if package_DIR is set. This patch prepends package_DIR (if specified) to the -L and -I results from the package's config tool instead of overriding them. Tested by: George Joseph Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3550/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b583f708cb	Fix build warnings with TEST_FRAMEWORK enabled ... ........ Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	874be8b530	Remove the problematic and unneeded AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c ... AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be incorrectly loaded before pbx_config. pbx_config was therefore blowing away contexts that were created by pbx_lua. With AST_MODFLAG_DEFAULT the load order is now correct and contexs are being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed anyway since no other modules needed its global symbols that early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by: Dennis Guse Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3629/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b960b9c2d5	Update extensions.lua.sample with naming conflict guidance. ... The sample extensions.lua was causing pbx_lua to fail to load when parsing 'app.goto("default", "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This patch adds guidance to extensions.lua.sample and changed 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	cc7bc40c2a	Allow the PUSH and UNSHIFT functions to set inheritable channel variables. ... ........ Merged revisions 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	15d2f541b4	MoH: Don't restart stream on repeated start calls ... Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	07a0838e2a	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db. ...	11 years ago
Matthew Jordan	8d848c048d	MoH: Undo commit r416150 (1.8) ... This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. ........ Merged revisions 416251 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	71c9c7612e	MoH: Don't restart stream on repeated start calls ... Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	11553fd489	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	7e2ed2e032	main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text ... Adds descriptive help text to 'core show hints' and 'core show hint'. The text describes the various columns for the sake of clarity. ASTERISK-23764 Review: https://reviewboard.asterisk.org/r/3610/ ........ Merged revisions 415998 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c7df3bd093	chan_sip: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero ... Change debug level for messages in sdp_crypto.c from zero to one. This ensures the messages are not displayed when debugging is disabled. Change does not apply to 12+ as it was already fixed in those versions. ASTERISK-23246 #close Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3605/ ........ Merged revisions 415908 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	df686c50d8	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	0df802b473	Correct UPGRADE.txt notes in r415825 ... The change was marked against the wrong version of Asterisk. My apologies. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	a5f39fc2ba	app_queue: delayed state can cause early leavewhenempty ringing ... In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont ........ Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	064bd035e7	MixMonitor: Add class authorization requirements to MixMonitor AMI commands ... MixMonitor AMI commands StartMixMonitor and StopMixMonitor lacked class authorization. StopMixMonitor now requires that the manager user either have the call or system class authorization. StartMixMonitor is a slightly larger issue since it can execute shell commands if the right arguments are passed into it, and we consider this a permission escalation. A security release will be issued for problem this shortly. ASTERISK-23609 #close Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	33a4ae86a1	format.c: Fix misuse of hash container function. ... The supplied hash function to a container must be idempotent given the object's key value to figure out which container bucket the object belongs in. Returning a random number or the current container count is not idempotent. The "computed hash" value doesn't help find the object later in those cases. * Fixed the format_list container to actually be a list since that is how the container is used. Conceptually, if more than 283 formats were added to the format_list then odd things may have happened before the fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	73945f9a97	chan_ooh323: fix loading module failure if there no accessible h323_log or ooh323 config file ... change return 1 to return AST_MODULE_LOAD_FAILURE on module load routine few cosmetic changes ASTERISK-23814 #close (closes issue ASTERISK-23814) Reported by: Igor Goncharovsky Patches: ASTERISK-23814-ast11.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	ea8009da8e	safe_asterisk: Cleanup additions to r415132. ... Replaced a stray echo that should've been a message call in safe_asterisk. I'm using the contents of the old message inside the if $NOTIFY so peoples log parsing scripts won't get confused by new messages. I'll clean that up in trunk. (Note that a 'make install' still won't overwrite your old safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 #close ........ Merged revisions 415521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2173e79720	autoservice: stop thread on graceful shutdown ... This change adds thread shutdown to autoservice for graceful shutdowns only. ast_register_cleanup is backported to 1.8 to allow this. The logger callid is also released on shutdown in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3594/ ........ Merged revisions 415463 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a92d272d2f	chan_sip: Fix order of variables specified in SIPNotify action ... Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4cc6c7683c	config: Fix config files not reloading when only an included file changes. ... The twisted logic determining if a config file should be reloaded was mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed actually reenabled the broken logic. The incorrect test was causing the timestamp to always be cleared which caused config files with includes to always be reloaded. * Made wildcard includes always cause a reload. Determining if a file was deleted cannot be determined without restructuring the cache to determine if any files are missing from the last files actually loaded. Also without refactoring config_text_file_load(), the glob loop couldn't check more than one file for changes anyway. * Made remove the cache entry if the file no longer exists when trying to get its timestamp or it is no longer a regular file. This fixes the corner case where the file was loaded, then deleted, then the config reloaded, then the file restored with the same timestamp, and then the config reloaded again. * Made remove the cache entry include list when actually loading the file. This gets rid of any stale includes the file had from the last time the file was loaded. ASTERISK-23683 #close Reported by: tootai Review: https://reviewboard.asterisk.org/r/3575/ ........ Merged revisions 415225 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a890f5469b	app_confbridge: Allow muting of users waiting to enter a ConfBridge ... Prior to this patch, users waiting to enter a ConfBridge were not considered when muted via the CLI or via AMI. Instead, a confusing message would be emitted stating that the channel did not exist. This patch allows a user to be muted when waiting to enter a ConfBridge conference. This is equivalent to start when muted, only toggled via the CLI or AMI. Review: https://reviewboard.asterisk.org/r/3582 ASTERISK-23824 #close patches: rb3582.patch uploaded by tm1000 (License 6524) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	2fbf3e8811	safe_asterisk: Cleanup and debian compatibility. ... Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND option that allows the debian asterisk init script to capture the right pid. * Drop the vim #modeline which wasn't used. Use test consistently without the odd configure xno syntax. Double quote all paths. General cleanup. * Don't output message()s to the console but only to TTY if set. * Allow TTY to be "no" as well as empty (debian compatibility with debian/patches/safe_asterisk-config). * Add option to export ASTSAFE_FOREGROUND=1 from the init script that calls this to disable backgrounding. Debian uses a similar method in debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: https://reviewboard.asterisk.org/r/3574/ ........ Merged revisions 415132 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	b9838a9960	app_confbridge: Correct verification of conference name length ... Conference names were not checked for maximum length, allowing unexpected behaviour. This change adds checking to ensure the maximum length is not exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909) confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	fb0754681a	func_odbc: Fix fixed size buffers fix (r414968). ... The change that removed the fixed size buffers in odbc-related code -- removing arbitrary column width limits -- was incomplete. This change adds: no segfault on writesql without insertsql and return value checks after strdup. While I was in the vicinity I cleaned up the linefeeds in the odbc function descriptions, moved some code for clarity, removed some blobs and noted (but didn't fix) that the 'odbc write ... exec' CLI command doesn't behave as the dialplan equivalent when insertsql= is used. #ASTERISK-23582 #close Review: https://reviewboard.asterisk.org/r/3579/ ........ Merged revisions 414997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c0494ad9c9	main/config.c: AMI action UpdateConfig EmptyCat clears all categories ... When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk will make all categories empty in the config but the one requested with a Cat variable. This is due to a bug in ast_category_empty (main/config.c) that makes an incorrect comparison for a category name. This patch corrects the comparison such that only the requested category is cleared. Review: https://reviewboard.asterisk.org/r/3573/ ASTERISK-23803 #close Reported by: zvision patches: manager.c.diff uploaded by zvision (License 5755) ........ Merged revisions 414880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b4042d348f	PBX: Prevent incorrect hint parsing ... Dynamic and pattern matching hints should not be checked for their last known state until they are instantiated by subscribers. (closes issue AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283) ........ Merged revisions 414813 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	80237dcf5b	res_config_odbc: Use dynamically sized buffers to store row data so values do not get truncated. ... ASTERISK-23582 #close ASTERISk-23582 #comment Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/3557/ ........ Merged revisions 414693 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	1ab8cca110	chan_unistim: Unlock mutex in rare OOM condition. ... ASTERISK-23792 #close Reported by: Peter Whisker Review: https://reviewboard.asterisk.org/r/3567/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	611f27fbd9	chan_sip: Start session timer at 200, not at INVITE. ... Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ ........ Merged revisions 414620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	7041eee5e5	res_config_odbc: Fix old and new ast_string_field memory leaks. ... The ODBC realtime driver uses ^NN parameter encoding to cope with the special meaning of the semi-colon. A semi-colon in a field is interpreted as if the key was supplied twice, something which isn't otherwise possible with fixed database columns. E.g. allow=alaw;ulaw is parsed as allow=alaw and allow=ulaw. A literal semi-colon is rewritten to ^3B when stored in the database. The module uses a stringfield to efficiently store the encoded parameters. However, this stringfield wasn't always freed in some off-nominal cases. Commit r413241 fixed initialization so the encoding for INSERT and DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked apparently.) But that commit forgot the frees. This change cleans that up. Review: https://reviewboard.asterisk.org/r/3555/ ........ Merged revisions 414564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	f992988b1f	Blocked revisions 414488 ... ........ Backport Asterisk 11 r413876 to 1.8 ........ r413876 | jrose | 2014-05-13 12:40:00 -0500 (Tue, 13 May 2014) | 6 lines chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c3d1e68377	app_meetme: Don't interrupt MOH for waitmarked users. ... Occasionally, when the last marked user leaves the conference, waitmarked users don't get MOH if MOH is supposed to be played while a waitmarked user is waiting for another marked user. * Made not interrupt MOH when the user is a waitmarked user. The waitmarked user doesn't need to hear any leave announcements from the conference as the user would have already heard different leave announcements if they were enabled. Apparently DAHDI occasionally sends unending non-silent streams to these users or a normal user still in the conference has continuous high background noise. These non-silent streams cause MOH to be suspended while the never ending "announcement" is played. Issue caused by ASTERISK-13680. AST-1349 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3543/ ........ Merged revisions 414401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	95eb7df060	UPGRADE: Add note for REF_DEBUG flag ... ........ Merged revisions 414345 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e5843ab97d	chan_local: Only block media frames when a generator is on both ends of a local channel. ... The fix for ASTERISK-12292 was a bit too aggressive. You could have generators pointed at each other on local channels but need to get other kinds of frames such as DTMF or CONNECTED_LINE frames accross. ........ Merged revisions 414269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	7d1a06a5dd	pbx.c: prevent potential crash from recursive replace() ... Recurisve usage of replace() resulted in corruption of the temporary string storage and potential crash. By changing the string to be allocated separtely per instance, this is eliminated. ASTERISK-23650 #comment Reported by: Roel van Meer ASTEIRSK-23650 #close Review: https://reviewboard.asterisk.org/r/3539/ ........ Merged revisions 414214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	f704632ffa	chan_ooh323: fix h323_log full path name ... * fix to use astlogdir option for h323_log file instead of hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky Patches: ooh323_logger_patch.diff ........ Merged revisions 414152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c9e1d7a154	chan_dahdi: Fix analog dialtone detection. ... * Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to allow the DSP to operate early enough to detect dialtone. * Made use the correct variable in my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve Davies Patches: dialtone_detect_fix (license #5012) patch uploaded by Steve Davies Review: https://reviewboard.asterisk.org/r/3534/ ........ Merged revisions 414067 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago