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2012-05-10Coverity Report: Fix issues for error type CHECKED_RETURN for coreJonathan Rose
(issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ ........ Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Coverity Report: Fix issues for error type UNINIT in Core supported modulesJonathan Rose
(issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Change comment to use local channel name designators in features.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Fix channel opaquification slip-up in r365477Matthew Jordan
Those channels are opaque now... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Support VoiceMail d() option when extension does not exist in channel's contextMatthew Jordan
The VoiceMail d([c]) option is documented to accept digits for a new extension in context <c>, if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the <c> option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 ........ Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365475 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04Fix many issues from the NULL_RETURNS Coverity reportKinsey Moore
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Multiple revisions 365006,365068Terry Wilson
........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365083 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Only log a failure to get read/write samples from factories if it didn't happenMatthew Jordan
In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ Merged revisions 364965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Fixed __ao2_ref() validating user_data twice.Richard Mudgett
(closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364903 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Prevent a potential crash when using manager hooks.Jason Parker
Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Fix bad check in voicemail functions for ast_inboxcount2_funcJonathan Rose
Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364777 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-30Merged revisions 364635 via svnmerge from Mark Murawki
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ Merged revisions 364650 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28md5: supress some compiler warnings.Russell Bryant
md5.c: In function ‘MD5Final’: md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] There is an md5 unit test and it still passes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28features: Add FEATURE() and FEATUREMAP() functions.Russell Bryant
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the [general] section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. See the built-in documentation for details. Review: https://reviewboard.asterisk.org/r/1871/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-27Multiple revisions 364365,364369Terry Wilson
........ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines Add missing test_config.c ........ Merged revisions 364365,364369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26Fix DTMF atxfer running h exten after the wrong bridge ends.Richard Mudgett
When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364065 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26Add more constness to the end_buf pointer in the netconsoleTerry Wilson
issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364047 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Make it possible to change the minimum DTMF duration in asterisk.confOlle Johansson
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Formatting fixesOlle Johansson
Developer guidelines are important. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Formatting fixesOlle Johansson
Found a small amount of curly brackets in my hotel room here in Denmark. I hereby donate them to the Asterisk project. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Fix recalled party B feature flags for a failed DTMF atxfer.Richard Mudgett
1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Hangup affected channel in error paths of bridge_call_thread().Richard Mudgett
........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363376 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-24OpenBSD doesn't have rawmemchr, use strchrTerry Wilson
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362869 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-23On some platforms, O_RDONLY is not a flag to be checked, but merely the ↵Tilghman Lesher
absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363212 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-23AST-2012-004: Fix an error that allows AMI users to run shell commands sans ↵Jonathan Rose
authorization. As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363156 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Fix connected-line/redirecting interception gosubs executing more than intended.Richard Mudgett
* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so execution will stop after the routine returns there. (s@gosub_virtual_context:1) * Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and gosub application respectively with the parameter string already created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Move debug message in ast_rtp_instance_early_bridge_make_compatible().Richard Mudgett
Move debug message in ast_rtp_instance_early_bridge_make_compatible() to be output when what it states has actually happened. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Add missing payload type to events APIMichael L. Young
The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362918 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Use ast_channel_lock_both() where it was inlined before.Richard Mudgett
The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel lock was originally obtained is overkill where ast_channel_lock_both() was inlined. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20* Add more information to some messages in __ast_pbx_run().Richard Mudgett
* Simplify some dialplan priority setting code in ast_explicit_goto() because of opaquification. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Add original party id and reason support.Richard Mudgett
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19Handle multiple commands per connection via netconsoleTerry Wilson
Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362537 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19Fix a variety of potential buffer overflowsMatthew Jordan
* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-17Handle case where an unknown format is used to get the preferred codec sizeMatthew Jordan
In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-17Fix places in main where a negative return value could impact executionMatthew Jordan
This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362360 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-17Avoid cppcheck warnings; removing unused vars and a bit of cleanup.Walter Doekes
Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16Add IPv6 address support to security events framework.Michael L. Young
The current Security Events Framework API only supports IPv4 when it comes to generating security events. This patch does the following: * Changes the Security Events Framework API to support IPV6 and updates the components that use this API. * Eliminates an error message that was being generated since the current implementation was treating an IPv6 socket address as if it was IPv4. * Some copyright dates were updated on files touched by this patch. (closes issue ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael L. Young Patches: security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12Convert SRV lookup message to debug levelPaul Belanger
This helps clean up the Asterisk CLI by converting the log message from verbose to debug git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12Add option to invoke the extensions.conf stdexten using the legacy macro method.Richard Mudgett
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-10Fix crash caused by unloading or reloading of res_http_postMatthew Jordan
When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. ........ Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361804 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist.Richard Mudgett
Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key header did not already exist. If it already existed it would delete it. * Made msg_set_var_full() exit early if the named variable did not already exist and the value to set is empty. ........ Merged revisions 361522 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06Add missing newlines to CLI loggingKinsey Moore
........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04Replace GNU old-style field designator extensions to fix clang warningsJonathan Rose
(issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) ........ Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8 Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue ASTERISK-19540) ........ Merged revisions 361143 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03Allow the Hangup manager action to match channels by regexMark Murawki
* Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-30Fix logger deadlock on Asterisk shutdown.Richard Mudgett
The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360934 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Fix potential race condition during call pickup.Mark Michelson
Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) ........ Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360885 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360886 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Misc changes to make astobj2 enhancement diffs easier to follow.Richard Mudgett
* Rename astobj2 API parameter funcname to func. * Rename astobj2 API iterator parameter to iter. * Update some documentation for OBJ_MULTIPLE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29undoing 360785 due to merging mistakeJonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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