asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	78089aa1cf	chan_sip: Allow static realtime members to be qualified during module load. When a static realtime peer with qualify=yes is loaded, Asterisk will fail to send an OPTIONS request due to the lastms being equal to 0. This results in the peer being unable to receive calls from Asterisk because the status is permanently UNKNOWN. This patch allows an OPTIONS request to be sent during module load by ignoring the lastms value on startup only. Review: https://reviewboard.asterisk.org/r/3294/ (closes issue ASTERISK-17523) Reported by: Maciej Krajewski Tested by: wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	ad637d43d5	moh: fix a refcount error with realtime MOH I observed a crash in res_musiconhold on an Asterisk 11 system using realtime MOH. Investigation of the backtrace showed a corrupt mohclass, implying that it got destroyed before the code expected it to. I went looking for reference counting errors that could have caused this crash and this patch this result. It contains 2 changes. 1) Remove a usless block of code that was impossible to reach. There was even a comment indicating that it was impossible to reach. The conditional includes "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if block with the opposite check "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no good reason to keep it around. 2) A similar block to #1 contained a reference counting error. It stores state->class in the local variable mohclass without increasing its reference count. The reference count on mohclass is decremented at the end of the function. This block of code probably very rarely runs, which would help explain why this system was working fine for many months before experiencing a crash. Review: https://reviewboard.asterisk.org/r/3282/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 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	11 years ago
Sean Bright	9808696b8a	Fix references to 'keys' CLI commands in astgenkey git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	28b726e79b	Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'	11 years ago
Michael L. Young	0e3e1b678c	func_audiohookinheritance: Check If A Channel Was Specified This patch prevents a crash when using the function audiohookinheritance without setting the channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by: Joel Vandal Patches: asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3272/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 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	11 years ago
Sean Bright	2942d6fdac	Minor whitespace change to 'sip show peers' output. (closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	c6f7a6652a	buildsystem: Unbreak 'make -qp' on 1.8. r408083 caused trouble with make -qp. Backport r408193 to 1.8 as well. (closes issue ASTERISK-23382) Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Tzafrir Cohen	25d1057845	Makefile: replace -O6 with -O3 -O6 is not a legal option of gcc. Unofficially gcc considers it to be equivalent of -O3. clang chalks on it, though. This commit sets the default optimization flag to be -O3, like gcc actually considered it. Review: https://reviewboard.asterisk.org/r/3280/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5e47d100d1	chan_sip: Add precautionary p->owner checks. * Add precautionary p->owner checks in sip_hangup(), get_refer_info(), get_also_info(), and interpret_t38_parameters(). * Simplify some tangled logic in get_refer_info(), get_also_info(), and add_rpid(). * Removed some dead code in handle_request_invite(). (closes issue ASTERISK-23323) Reported by: Walter Doekes Patches: issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5f27cde1ea	chan_sip: Fix crash in ast_channel_hangupcause_set(). * Fix crash in ast_channel_hangupcause_set() because p->owner not checked before calling. Regression introduced by the fix for ASTERISK-22621. (closes issue ASTERISK-23135) Reported by: OK (issue ASTERISK-23323) Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	b695da6510	Fix memory stomping bug in astman. This memset complained in dev mod on my Ubuntu box. The memset is both unnecessary and dangerous. At this point, m hasn't been initialized yet, so the memset will write off to whatever address happens to be on the stack at the time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a6a92ffe3d	res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load. Change minrate from 2400 to 4800 on config reload in response to changes from ASTERISK-22790 only. Any config with minrate of 2400 that would fail before r405693 will still fail. Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Rusty Newton	d72f70e9a5	configs/voicemail.conf.sample - Make mailcmd sample text more explicit Made the wording a bit more explicit. Didn't really change the meaning. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Kevin Harwell	c8e50614fb	app_forkcdr: ForkCDR v option does not keep CDR variables for subsequent records When the 'v' option is specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is set only for the first CDR in the chain. So ForkCDR works fine with this option only once. After the second and further calls to ForkCDR, CDR variables get cleared on all CDRs besides the first one and moved to the newly forked CDR. It always sets the KEEP_VARS flag on the first CDR in the chain, instead of the most recent CDR which is used as a base to fork a new CDR. This patch sets KEEP_VARS flag on the most recent CDR on the stack (the CDR used for forking). (closes issue ASTERISK-23260) Reported by: zvision Patches: app_forkcdr.diff uploaded by zvision (license 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 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	11 years ago
Alexandr Anikin	44af8d9f46	Fix type of roundTripDelay variables git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	2b4a90343a	app_chanspy: Documentation Update To Clarify "x" Option When using the "x" option (specify a DTMF digit to exit the application), it is not obvious in the documentation that this only works when spying on a channel. If a channel being used to spy on other channels is waiting to connect to a channel or is no longer attached to a channel, the DTMF is ignored. As noted on the issue tracker, since there are workarounds available and this is a rarely used option we are opting for a documentation change here. (closes issue ASTERISK-22661) Reported by: Chris Hillman Patches: asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2990/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	8c1e17b171	apps/app_queue - Fix incorrect Macro parameter documentation Macro is executed on the called channel, not the calling channel. (closes issue ASTERISK-23069) Reported By: Bryan Anderson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Alexandr Anikin	8c234f205d	process receiveAndTransmit user input remote caps instead of receive only send receiveAndTransmit user input our caps instead of receive only git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 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	11 years ago
Walter Doekes	b78a146975	buildsystem: Don't force main to depend on everything else. Directory 'main' only needs to depend on embedded modules. If no module embedding is selected, the dependency is dropped. Review: https://reviewboard.asterisk.org/r/3212/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	f0f80e418b	configs/agents.conf.sample - Remove example for non-functional "goodbye" parameter The "goodbye" parameter is not implemented in the source code, it does nothing. (closes issue SWP-6518) Reported By: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	f6aed66f77	res_config_pgsql: Fix ast_update2_realtime calls. Fix so multiple updates from a single call works (add missing ','). Remove bogus ast_free's that weren't supposed to be there. Moved a few spaces for readability. Review: https://reviewboard.asterisk.org/r/3194/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	8ee947ed5e	chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel When a DAHDI device is removed at run-time it sends the event DAHDI_EVENT_REMOVED on each channel. This is intended to signal the userspace program to close the respective file handle, as the driver of the device will need all of them closed to properly clean-up. This event has long since been handled in chan_dahdi (chan_zap at the time). However the event that is sent on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This commit adds handling for closing the file descriptor (and shutting down the span, while we're at it). It also adds a CLI command 'pri destroy span <N>' to destroy the span and its DAHDI channels. Backported from trunk/12. Review: https://reviewboard.asterisk.org/r/726/ ........ Merged revisions 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	414d336f45	chan_iax2: Add some more iaxs[] NULL checks to a routine already full of them. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6e6a7f2af0	chan_iax2: Block unnecessary control frames to/from the wire. Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	8878f38742	indications.conf: add stutter tone; end properly * If the "stutter" (voicemail indication) tone is indeed a stutter tone, and it ends with a constant tone, make sure that it is the dial tone. This was done for India (in), Mexico (mx) and the Philippines (ph). * If no "stutter" tone exists for a country, provide one. This was done for Spain (es), Malaysia (my) and Venezuela (ve). Review: https://reviewboard.asterisk.org/r/3158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	c31d664b0a	formats/format_wav: enhancing log message "Not a wav file" to be clear on what is supported Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension. (closes issues ASTERISK-22310) Reported by: Jim Credland Review: https://reviewboard.asterisk.org/r/3188/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Richard Mudgett	d1b9b7fc77	devicestate: Make ast_devstate_changed_literal() return value and doxygen consistent. Nothing actually cares about the value anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	d06ec403aa	tcptls.c: Made TLS handle a certificate chain file. Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3e1ec590f3	res_clialiases: Fix crash when reloading and re-aliasing an alias that is in use. The code assumed that unregistering the alias would always succeed while in practice this is not actually true. A common case is the "reload" command itself. If the cli_aliases.conf configuration file was changed and reload executed the command would fail to unregister and ultimately point to freed memory. The reload process now checks whether unregistering succeeded or not and if not the old CLI alias is retained. (closes issue ASTERISK-19773) Reported by: Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth Blades git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	36771d3004	app_stack: protect against missing parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This protects against situations where those parameters are blank or missing by logging an error and returning. (closes issue ASTERISK-23220) Reported by: James Sharp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c5e4585565	app_dial: Allow macro/gosub pre-bridge execution to occur on priorities The parsing for the destination of the macro/gosub uses the '^' character to separate out context, extension, and priority. However, the logic for the macro/gosub execution was written such that it would only do the actual macro/gosub jump if a '^' character existed. This doesn't apply when the macro/gosub jump occurs in a priority/priority label. This patch changes the logic so that the parsing still occurs, but the jump will occur even for priorities/priority labels. (issue ASTERISK-23164) Review: https://reviewboard.asterisk.org/r/3154 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ea65e4d44c	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	f82e6e93c5	queues.conf.sample Fix documented default for persistentmembers Closes issue ASTERISK-22662 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	88794101a9	cdr_radius, cel_radius: build agains libfreeradius-client Asterisk's RADIUS module currently build against libradiusclient-ng, but this project has been superseeded by libfreeradius-client. The API is 99% compatible except that the header name has changed, the library name has changed, and the configuration file location has changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé Patches: freeradius-client.patch uploaded by sharky (license 6561) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	9ab9f6e6b4	rtp_engine: improved handling of get_rtp_info failure In ast_rtp_instance_make_compatible(), after a failure of channel tech call get_rtp_info() to return peer_instance, the null pointer would be passed to ao2_ref, producing an error that looked like a refernce counting problem but is not. This patch corrects that and adds helpful LOG_ERROR messages to indicate which failure path occurred. (issue AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	bfd09ecfd4	Allow nested #includes in extconfig.conf extconfig.conf was hard-coded to not allow nested includes for some reason. The code has been this way since a patch was merged for ASTERISK-3333 (revision 4889), which was a significant update to this code ("Merge config updates"). I can't figure out any good reason why this should be limited. This patch just removes the limit and uses the default nesting depth limit. Closes issue ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	270775ce52	Protect ast_filestream object when on a channel The ast_filestream object gets tacked on to a channel via chan->timingdata. It's a reference counted object, but the reference count isn't used when putting it on a channel. It's theoretically possible for another thread to interfere with the channel while it's unlocked and cause the filestream to get destroyed. Use the astobj2 reference count to make sure that as long as this code path is holding on the ast_filestream and passing it into the file.c playback code, that it knows it's valid. Bug reported by Leif Madsen. Review: https://reviewboard.asterisk.org/r/3135/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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