3923 Commits (cc46ce38bc98e1544d6d077c910238ee0c7b51ed)

Author	SHA1	Message	Date
Corey Farrell	8745e12323	main/file.c: fix possible extra ast_module_unref to format modules. ... fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e9f3480121	Fix unintential memory retention in stringfields. ... * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	03fcc1ad72	config: Make text_file_save and 'dialplan save' escape semicolons in values. ... When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9dc2f92921	Fix ast_writestream leaks ... Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0b55748232	func_jitterbuffer: fix frame leaks. ... Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0a2400ab6e	main/manager: Fix typo in AMI event documentation of "OriginateResponse" ... The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8c96fa291c	AST-2014-011: Fix POODLE security issues ... There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all encryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e6bd251926	CallerID: Fix parsing regression ... This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5454c620e0	astobj2: Correct REF_DEBUG false leak report ... When ao2_callback is run with OBJ_MULTIPLE and not OBJ_NODATA it allocates a temporary container in a way that does not record REF_DEBUG log entries. This changes that container to correctly record unref's when the container is freed. ASTERISK-24390 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4047/ ........ Merged revisions 424786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	478e64f1e4	message: Don't close an AMI connection on SendMessage action error ... If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7d06282b9e	Release AMI connections on shutdown. ... ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a1fab60fcc	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. ... r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7d0f1a26e6	astobj2.c/refcounter.py: Fix to deal with invalid object refs. ... * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c4cf004f24	config: bug: Fix SEGV in ast_category_insert when matching category isn't found ... If you call ast_category_insert with a match category that doesn't exist, the list traverse runs out of 'next' categories and you get a SEGV. This patch adds check for the end-of-list condition and changes the signature to return an int for success/failure indication instead of a void. The only consumer of this function is manager and it was also changed to use the return value. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3993/ ........ Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6badd734d3	Bridging: Fix bouncing native bridge ... This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge could cause a bouncing native bridge. In the case of the dial_LS_options test, this was a remote RTP bridge which caused the audio path to continually cycle between Asterisk and the remote endpoints generating a large number of SIP messages and delaying the test long enough to cause it to fail (checking timing was part of the test). The root cause was that the code to decide whether to use native bridging was expecting a time-remaining value of 0 to be the default instead of the actual default value of -1. A value of 0 or negative numbers could also be generated by preceding code in some circumstances. Both issues are addressed in this patch. ASTERISK-24211 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3987/ ........ Merged revisions 423006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	2fbdeafb19	config: bug: fix truncation of included config files on permissions error ... ast_config_text_file_save() currently truncates include files as they are processed. If a subsequent include file or the main config file has a permissions error that prevents writing, earlier include files are left truncated resulting in a frantic search for backups. This patch causes ast_config_text_file_save to check for write access on all files before it truncates any of them. Will be applied 1.8 > trunk. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3986/ ........ Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	388a13abb8	Manager: Require read permission for SYSTEM in order to send FullyBooted ... Review: https://reviewboard.asterisk.org/r/3969/ ........ Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc3fb5bcb6	manager: Make WaitEvent action respect eventfilters ... A WaitEvent issued via an http session isn't respecting eventfilters defined for the user. I just added a match_filter to the predicate that controls astman_append. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3958/ ........ Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5cdd1413f1	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/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	05c31108a5	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(). ........ Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	16a3e1cc3f	AMI Docs: Fix Status channel parameter optionality ... ........ Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	252ead3b13	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. ........ Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	174c84f6a2	main/file: Move test event to emit PLAYBACK event more consistently ... This is being done in advance of the test for ASTERISK-23953 ........ Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	a63e70c46d	logger: Don't store verbose-magic in the log files. ... In r399267, the verbose2magic stuff was edited. This time it results in magic characters in the log files for multiline messages. In trunk (and 13) this was fixed by the "stripping" of those characters from multiline messages (in r414798). This is a backport of that fix to 11. That fix is altered to actually strip the characters and not replace them with blanks. Review: https://reviewboard.asterisk.org/r/3901/ Review: https://reviewboard.asterisk.org/r/3902/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	9aba91e5a6	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/ ........ Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	d43da8846c	tcptls: Avoid compiler warning on non-dev-mode. ... ........ Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	0430c887f3	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/ ........ Merged revisions 420146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9707876002	format.c: Add reason comments for the format_list ordering. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	af592bbc94	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 ........ Merged revisions 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4c47641f0c	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/ ........ Merged revisions 419630 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bfc6904871	manager: Return ActionID on nominal responses to PresenceState action ... When the PresenceState action is executed, the nominal path fails to include the ActionID in the successful response. This patch adds a call to astman_start_ack, which guarantees that an ActionID (if provided) will be sent back to the AMI client. Review: https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	148b8d128e	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/ ........ Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	a100307eaa	config: inform config hook of change when writing file ... When updated configuration is written back to the conf file - for example when a user changes their voicemail pin, make sure that any config hook that wants to know of changes is informed. Review: https://reviewboard.asterisk.org/r/3708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8f5461d3dc	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) ........ Merged revisions 417797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	915de454f8	res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. ... This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 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	12 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. ...	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago
Walter Doekes	4471447ea8	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/ ........ Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	abac3330cf	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. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago