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Author	SHA1	Message	Date
Walter Doekes	edca0bd412	tcptls: Avoid compiler warning on non-dev-mode. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@420654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7153c2c2c5	chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue. ... Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@420434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	76d1ae97d4	pbx_lua: fix regression with global sym export and context clash by pbx_config. ... ASTERISK-23818 (lua contexts being overwritten by contexts of the same name in pbx_config) surfaced because pbx_lua, having the AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before pbx_config. Since I couldn't find any reason for pbx_lua to export it's symbols to the rest of Asterisk, I simply changed the flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't realize was that the symbols need to be exported not because Asterisk needs them but because any external Lua modules like luasql.mysql need the base Lua language APIs exported (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's an issue in pbx.c where context_merge was only merging includes, switches and ignore patterns if the context was already existing AND has extensions, or if the context was brand new. If pbx_lua is loaded before pbx_config, the context will exist BUT pbx_lua, being implemented as a switch, will never place extensions in it, just the switch statement. The result is that when pbx_config loads, it never merges the switch statement created by pbx_lua into the final context. This patch sets pbx_lua's modflag back to AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge that catches the case where an existing context has includes, switchs or ingore patterns but no actual extensions. ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo Teräs Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@420146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	e3ad222bb0	Manager - Improve documentation for manager commands Getvar and Setvar. ... The documentation for these commands did not make it clear that they could accept expressions and functions. Modified to make this clear, but tried not to be overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3854 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	dae78b0892	datastores: Audit ast_channel_datastore_remove usage. ... Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in app_speech_utils and func_frame_trace. * Fixed app_speech_utils not locking the channel when accessing the channel datastore list. Review: https://reviewboard.asterisk.org/r/3859/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9e3411a762	features.c: Allow appliationmap to use Gosub. ... Using DYNAMIC_FEATURES with a Gosub application as the mapped application does not work. It does not work because Gosub just pushes the current dialplan context, exten, and priority onto a stack and sets the specified Gosub location. Gosub does not have a dialplan execution loop to run dialplan like Macro. * Made the DYNAMIC_FEATURES application mapping feature call ast_app_exec_macro() and ast_app_exec_sub() for the Macro and Gosub applications respectively. * Backported ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute dialplan routines from the DYNAMIC_FEATURES application mapping feature. NOTE: This issue does not affect v12+ because it already does what this patch implements. AST-1391 #close Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3844/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0789fefda6	chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy ... sip_subscribe_mwi_destroy calls sip_destroy on the reference counted mwi->call. This results in the fields of mwi->call being freed, but mwi->call itself it leaked. If other code is still using mwi->call it can cause problems. This change uses dialog_unref instead, to balance the ref provided by sip_alloc(). ASTERISK-24087 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	94afd8c438	Don't cause Asterisk to exit if ooh323.conf not found. ... (closes issue ASTERISK-23814) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	912f2287e9	Fix more dev-mode build issues ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	3ee4ecb045	func_uri: URIENCODE/URIDECODE - allow empty strings as argument ... Previously these two dialplan functions would issue warnings and return failure when an empty string is used as the argument. Now they will not issue a warning and will successfully return an empty string. ASTERISK-23911 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3745/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@418641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	01492e96d9	astobj2: work around REF_DEBUG race which causes out of order log entries ... * Update refcounter.py to use delta's to track the current reference count. * Use result from internal_ao2_ref to write old_refcount to refs_log. Review: https://reviewboard.asterisk.org/r/3756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@418504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4542181b8a	chan_dahdi/sig_pri: Fix type mismatch in the idledial feature's channel creation. ... Square pegs in round holes don't work very well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@418261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	863e7e528c	chan_dahdi: Add inband_on_setup_ack compatibility option. ... The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	904a6b3910	main/untils: Prevent potential infinite loop in ast_careful_fwrite ... A loop in ast_careful_fwrite exists that will continually attempt to write to a file stream, even in the presence of EAGAIN/EINTR errors. However, if a connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to fflush will return EOF but not clear errno, resulting in an infinite loop. This patch clears errno after it is detected and handled the loop, such that any subsequent call to fflush will not get erroneously stuck. Review: https://reviewboard.asterisk.org/r/3704 ASTERISK-23984 #close Reported by: Steve Davies patches: fflush_loop_fix uploaded by one47 (License 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	67a11896e7	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. Note that this patch only applies to audio in Asterisk 1.8, which is a bit more limited in its support for format attributes. It does have limited support for some codecs, so this patch is still useful in this version. Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5386eeda30	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d8a04cd816	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	db3d055e38	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c56504a148	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	d3d407c175	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 #close Review: https://reviewboard.asterisk.org/r/3621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c64fde05ab	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	356a1f8620	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	46c8b1eefd	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/1.8@416869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7752d56964	Fix build warnings with TEST_FRAMEWORK enabled ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8184532a9f	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/1.8@416667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c2c3e17138	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)'. https://reviewboard.asterisk.org/r/3627/ ASTERISK-23844 #comment This commit fixes 1.8, patch for 11->trunk coming. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	964fac8524	Allow the PUSH and UNSHIFT functions to set inheritable channel variables. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	21c4871730	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	b718f91873	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	6b782f84b8	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4707dd1ce5	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1509ae5779	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	989b441f6c	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	883d2e6924	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7550fe7efb	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	855fceccac	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	9972136127	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 ASTERISK-23814 #close (closes issue ASTERISK-23814) Reported by: Igor Goncharovsky Patches: ASTERISK-23814-ast18.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	ef59e04d6c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	1dff17f051	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	560380a018	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ae71ca1d2f	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	14cfff4a02	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	45c62c1733	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	6175902998	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c1be1f43cb	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	27d80767db	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	8a52013dca	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	bf94ff5763	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	81b4fbb578	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	71df5a4a63	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/1.8@414488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago