asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	54bf60c118	CallerID: Fix parsing of malformed callerid This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@422112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f92b7fe87d	cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled. filename_completion_function() returns memory that was not allocated by the MALLOC_DEBUG allocation tracker so the memory must be freed by ast_std_free(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0b36c21dfd	AMI Docs: Fix Status channel parameter optionality git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5ee6da4cf2	app_voicemail/app: Remove test events that were duplicated by r421059 Moving the test event raised when a file is played back (which occurred in r421059) broke the ever loving snot out of the voicemail tests. This caused duplicate test events to get raised, as app_voicemail and main/app were raising events prior to call ast_streamfile. The voicemail tests did not enjoy getting multiple events. Since raising the playback event in ast_streamfile is far more useful to the vast majority of tests, this patch keeps the call there and simply removes the extraneous calls that duplicated the event. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9c477681f1	main/file: Move test event to emit PLAYBACK event more consistently This is being done in advance of the test for ASTERISK-23953 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	6d6c119777	general: Fix memory Corruption in __ast_string_field_ptr_build_va. If the space left in a stringfield is between 0 and (alignof(ast_string_field_allocation)-1) adding new data would cause memory corruption, because we would assume enough space (unsigned underrun). Thanks Arnd Schmitter for reporting and finding out the cause! ASTERISK-23508 #close Reported by: Arnd Schmitter Tested by: Arnd Schmitter, JoshE Review: https://reviewboard.asterisk.org/r/3898/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@420680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago
Walter Doekes	228d5c9052	rtp: Fix case typo in H263+ mime. http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical mime subtype is "H263-1998", not "h263-1998". Original code was added in r183101 on 2009-03-19 02:26:50 +0100. This fixes issues with Polycom phones. ASTERISK-23665 #close ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me. Review: https://reviewboard.asterisk.org/r/3529/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	3e9a54d857	Allow Asterisk to compile under GCC 4.10 This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cb8877e720	app_queue: Extend documentation for various Manager actions and events. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8ccad4ce21	http: Fix spurious ERROR message in responses with no content. Backport -r411687 and fix the fix because content_length is the length of out plus the length of the file controlled by fd. When a response has an out content length of 0, fwrite would be called to write a buffer with no data in it. This resulted in the following classic error message: [Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success This patch makes it so that we only attempt to write the content of out if the out string is non-zero. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	24f054f497	HTTP: Add TCP_NODELAY to accepted connections This adds the TCP_NODELAY option to accepted connections on the HTTP server built into Asterisk. This option disables the Nagle algorithm which controls queueing of outbound data and in some cases can cause delays on receipt of response by the client due to how the Nagle algorithm interacts with TCP delayed ACK. This option is already set on all non-HTTP AMI connections and this change would cover standard HTTP requests, manager HTTP connections, and ARI HTTP requests and websockets in Asterisk 12+ along with any future use of the HTTP server. Review: https://reviewboard.asterisk.org/r/3466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	751dc2b7da	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. Review: https://reviewboard.asterisk.org/r/3377/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	85a69184c7	Internal timing: Add notice that the -I and internal_timing option are no longer needed. Add notice messages during execution that the -I command line option and the astersik.conf internal_timing option are no longer needed. The internal timing functionality is now always enabled if there is a timing module loaded. NOTE: Since the command line options and the asterisk.conf config file are processed before the logging system is initialized, the messages are output to stderr. Change requested as a result of asterisk-dev list comments about the commit for ASTERISK-22846 that removed the -I and internal_timing options. Review: https://reviewboard.asterisk.org/r/3423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	70cdee628c	config: Fix CB_ADD_LEN() to work as originally intended. Fix a long standing bug in CB_ADD_LEN() behaving like CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a997ddc829	internal_timing: Remove the option and always make it enabled if a timing module is loaded. The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	dd5c0ffd4d	http: response body often missing after specific request This patch works around a problem with the HTTP body being dropped from the response to a specific client and under specific circumstances: a) Client request comes from node.js user agent "Shred" via use of swagger-client library. b) Asterisk and Client are not on the same host or TCP/IP stack In testing this problem, it has been determined that the write of the HTTP body is lost, even if the data is written using low level write function. The only solution found is to instruct the TCP stack with the shutdown function to flush the last write and finish the transmission. See review for more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) Reported by: Sam Galarneau Review: https://reviewboard.asterisk.org/r/3402/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e86e40892c	say: Fix a bug where SayNumber in Polish tries to play incorrect sound. This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russ Meyerriecks	e62a5f78e9	!fixup: callerid: Logic error in checksum processing Fixes syntax error in previous commit :-( git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russ Meyerriecks	2478dc0606	callerid: Logic error in checksum processing Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1d8a661b30	AST-2014-001: Stack overflow in HTTP processing of Cookie headers. Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	992b0aa702	config: Fix inverted test The test of the result of the stat() call was inverted such that its output was only used if the call failed. This inverts the test so that the output of stat() is used correctly. This was causing full reloads on unchanged files. (closes issue ASTERISK-23383) Reported by: David Woolley git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a15bacd163	Corrected cross-platform stat nanosecond code When nanosecond time resolution was added for identifying config file changes, it didn't cover all of the myriad of ways that one might obtain nanosecond time resolution off of struct stat. Rather than complicate the #if even further figuring out one system from the next, this patch directly tests for the three struct members I know about today, and #ifdef's accordingly. Review: https://reviewboard.asterisk.org/r/3273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	f75aeea441	AO2: Add an assert for bad objects This adds an assert that will only be active if Asterisk is compiled with DO_CRASH and allows the testsuite to fail tests that would otherwise require log file parsing. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	133a042f25	rtp_engine: Clean up after a failed remote bridge Upon failure of an INVITE transaction meant to initiate a remote native bridge, rtp_engine.c would not clean up non-reference-counted bridge instance pointers leaving a dangling pointer which was being used to perform a local native bridge after the other channel had hung up. This lead to dereferencing into freed memory and plenty of AO2 errors. This change allows the remote native bridge loop to clean up properly when the bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy Laine git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b42ab7fe81	doxygen: Tweak the link back to ye olde Digium website git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bb9dfa1962	rtp_engine: fix crash during remote native bridging when calling get_codecs When two RTP channels are in a remote bridge, the remote bridging loop in rtp_engine will periodically check to see if the two channels can still be bridged. One of the many things it checks is whether or not the codecs have changed on the channel. If the codec has changed, it will break out of the loop to re-determine which type of bridge is appropriate. In order to perform this check, the ast_rtp_glue virtual table's get_codec callback is called for each channel. The callback implementations assume that the channel tech private is valid when called; as such, there has always been some code in place to check whether or not the channel pvt is NULL before calling. However, this check is insufficient. The channels are unlocked during the remote bridging loop. It is possible for a channel to get masqueraded between the check for the pvt being NULL and the actual call to get_codec. When this occurs, the callback is called with a ZOMBIE channel, which now has a NULL pvt. Crash. While this has always been possible in Asterisk 1.8, it is much more likely to occur in Asterisk 11 and later versions due to the timing changes that occur when getting the codec from a channel. Note that this is much more likely to be reproduced on slow, boggy hardware running Asterisk 11 - but fairly rarely otherwise. Also Note: This crash was also caught by the various SIP blind transfer tests, in addition to the bug report Alec filed. Review: https://reviewboard.asterisk.org/r/3247/ (closes issue ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	b78f5a2d30	Remove extra defines of AST_PBX_MAX_STACK. * Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	d2e5c5f922	rtp_engine: Output mixup in ${CHANNEL(rtpqos,audio,all)} Fixed the output of CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter. (closes issue ASTERISK-23261) Reported by: rsw686 Patches: rtpqos.patch uploaded by rsw686 (license 5887) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2197b1d114	channel.c: MOH is not working for transferee after attended transfer Updated the code to check to see if MOH is playing on the transferor and if so then start it on the channel that replaces it during a masquerade. Example scenario of the problem: Alice calls Bob and then Bob begins the attended transfer process into a queue. Upon going on hold Alice hears music and so does Bob once he is in the queue. Bob then transfers Alice into the queue and then music for Alice stops even though she should be hearing it since has now replaced Bob in the queue. The problem that was occurring is that once the channel was masqueraded the app (queues, confbridge, etc...) had no way of knowing that the channel had just been swapped out thus it did not start music for the present channel. Credit to Olle Johansson for pointing me in the right direction on this issue. (closes issue ASTERISK-19499) Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/3226/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ba39ba7635	config: Add file size and nanosecond resolution fields to the cached modified config file information. Repeatedly modifying config files and reloading too fast sometimes fails to reload the configuration because the cached modification timestamp has one second resolution. * Added file size and nanosecond resolution fields to the cached config file modification timestamp information. Now if the file size changes or the file system supports nanosecond resolution the modified file has a better chance of being detected for reload. * Added a missing unlock in an off-nominal code path. (closes issue AST-1303) Review: https://reviewboard.asterisk.org/r/3235/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	834f3215a9	pbx: Handle a completely empty dialplan during a context merge It is highly unlikely, but - at least in Asterisk 12 - theoretically possible to load Asterisk with no dialplan whatsoever. If that occurs, and some other module (that is not a pbx module) attempts to merge its contexts into the dialplan, the existing merge routine will crash. This is because it is not insane, and rightly believes that you provided some sort of dialplan, somewhere. This patch will gracefully merge the contexts in such a case. Note that this is highly unlikely to occur in 1.8/11, as features will most likely provide some dialplan via parking. However, in Asterisk 12, parking is now provided by res_parking, and hence may create its dialplan later. (closes issue ASTERISK-23297) Reported by: CJ Oster Review: https://reviewboard.asterisk.org/r/3222 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	e0bed661dd	pbx: ast_custom_function_unregister resource leak In pbx.c ast_custom_function_unregister(), a list of escalations being removed from the list wasn't being free'd creating a leak. This patch corrects that by freeing the records. Review: https://reviewboard.asterisk.org/r/3213/ Reported by: Corey Farrell Patches: acf_escalating_leak.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d824987e9d	Logger: Fix handling of absolute paths This fixes path handling for log files so that an extra / is not appended to the file path when the path is absolute (begins with /). This would previously result in different but functionally equivalent paths in the output of 'logger show channels'. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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