asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	ab674f67b5	app_confbridge: file playback blocks dtmf Attempting to execute DTMF in a confbridge while file playback (prompt, announcement, etc) is occurring is not allowed. You have to wait until the sound file has completed before entering DTMF. This patch fixes it so that app_confbridge now monitors for dtmf key presses during menu driven file playback. If a key is pressed playback stops and it executes the matched menu option. ASTERISK-24864 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4510/ ........ Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433446 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e953d15223	A couple minor cleanup tweaks. * In res/res_sorcery_realtime.c: Broke long line. * In main/bucket.c: Eliminated unnecessary NULL check as ast_sorcery_unref() is NULL tolerant and set the global object to NULL after unref in the system shutdown bucket_cleanup(). ........ Merged revisions 433420 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	47156aab92	res_xmpp: Buddies are always auto-registered when processing the roster Due to a quirk in the configuration handling of res_xmpp, the 'autoregister' setting was never actually processed. This was due to not properly copying over the global settings to the client settings when applying the configuration to the run-time object. Review: https://reviewboard.asterisk.org/r/4496/ ASTERISK-14233 ASTERISK-24780 #close Reported by: Simon Arlott patches: asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756) ........ Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433396 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	abf3e40902	dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. This change adds an abstracted core DNS API which resembles the API described here[1]. The API provides a pluggable mechanism for resolvers and also a consistent view for records. Both synchronous and asynchronous queries are supported. This change also adds a res_resolver_unbound module which uses the libunbound library to provide resolution. Unit tests have also been written for all of the above to confirm the API and functionality. ASTERISK-24834 #close Reported by: Matt Jordan ASTERISK-24836 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4474/ Review: https://reviewboard.asterisk.org/r/4512/ [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4c2fc5b811	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ ........ Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	60f01520e7	Fix compilations errors on 64-bit OpenBSD systems In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to (long) when printing members of certain time structs. Review: https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close Reported by: snuffy Tested by: snuffy patches: openbsd-time64.diff uploaded by snuffy (License 5024) ........ Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433269 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	66670f02e6	Fix compilation issues for OpenBSD This patch addresses compilation issues for OpenBSD. Specifically, it addresses: * It allows including <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast in xmldoc.c In 13+, it also addresses a conditional inclusion in loader.c. Review: https://reviewboard.asterisk.org/r/4506 ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy patches: misc-openbsd.diff uploaded by snuffy (License 5024) ........ Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433247 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	7e097bce86	Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind found some memory leaks associated with ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending responses to OPTIONS requests, processing MESSAGE requests, and res_pjsip supplements implementing the incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(), res/res_pjsip/pjsip_options.c:send_options_response(), res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and res/res_pjsip_messaging.c:send_response(). * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but benign return value in res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review: https://reviewboard.asterisk.org/r/4511/ ........ Merged revisions 433222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	148e8799fe	res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively. Valgrind found a memory leak and invalid access. * Fix invalid access by sscanf() being fed a non-nul terminated string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix memory leak in main/sorcery.c:sorcery_object_field_destructor(). * Fix potential NULL pointer dereference in main/xmldoc.c:xmldoc_get_syntax_config_option(). Review: https://reviewboard.asterisk.org/r/4513/ ........ Merged revisions 433199 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	627cc16a8d	funcs/func_env: Fix regression caused in FILE read operation When r432935 was merged, it did correctly fix a situation where a FILE read operation on the middle of a file buffer would not read the requested length in the parameters passed to the FILE function. Unfortunately, it would also allow the FILE function to append more bytes than what was available in the buffer if the length exceeded the end of the buffer length. This patch takes the minimum of the remaining bytes in the buffer along with the calculated length to append provided by the original patch, and uses that as the length to append in the return result. This patch also updates the unit tests with the scenarios that were originally pointed out in ASTERISK-21765 that the original implementation treated incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433174 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	79a81fed59	alemebic scripts: endpoint identifier order option The script was added in 13, but when committed to trunk it caused a branch to occur due to some trunk only alemebic changes. This fixes it so that the new 'add_pjsip_endpoint_identifier_order script points to the correct down revision. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	3aa0a869c2	logger: Apply default console logging when configuration cannot be loaded. When logger.conf is missing or invalid enable console logging and display an error message. ASTERISK-24817 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4497/ ........ Merged revisions 433122 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433126 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	d486659502	chan_sip: Simplify dialog/peer references, improve REF_DEBUG output. * Replace functions for ref/undef of dialogs and peers with macro's to call ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller information to sip_alloc and find_call. ASTERISK-24882 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4189/ ........ Merged revisions 433115 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	2c83ac4364	chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout. Release the scheduler reference to the dialog for reinvite timeout during dialog_unlink_all. ASTERISK-24876 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4491/ ........ Merged revisions 433112 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433113 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e0ea490a11	res_pjsip_session: Fix off-nominal extra unref of session. ........ Merged revisions 433088 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	8c65c9167e	Various: bugfixes found via chaos Using DEBUG_CHAOS several instances of a null pointer crash, and one uninitialized variable were uncovered and fixed. Also added details on why Asterisk failed to initialize. Review: https://reviewboard.asterisk.org/r/4468/ ........ Merged revisions 433064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	f25b265329	core: Introduce chaos into memory allocations Locate potential crashes by exercising seldom used code paths. This patch introduces a new define DEBUG_CHAOS, and mechanism to randomly return an error condition from functions that will seldom do so. Functions that handle the allocation of memory get the first treatment. Review: https://reviewboard.asterisk.org/r/4463/ ........ Merged revisions 433060 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	62cf2a2c02	Reverting accidental ci of wrong change in r433061 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	cb6c7eecfd	various: cleanup issues found during leak hunt In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ ........ Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	c41dd32b94	Audit ast_sockaddr_resolve() usage for memory leaks. Valgrind found some memory leaks associated with ast_sockaddr_resolve(). Most of the leaks had already been fixed by earlier memory leak hunt patches. This patch performs an audit of ast_sockaddr_resolve() and found one more. * Fix ast_sockaddr_resolve() memory leak in apps/app_externalivr.c:app_exec(). * Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs parameter for safety so the pointer will never be uninitialized on return. The same goes for res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs variable use ast_free instead of ast_free_ptr to provide better MALLOC_DEBUG information. Review: https://reviewboard.asterisk.org/r/4509/ ........ Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433057 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	803a916334	res_pjsip: Allow configuration of endpoint identifier query order Updated some documentation stating that endpoint identifiers registered without a name are place at the front of the lookup list. Also renamed register method 'ast_sip_register_endpoint_identifier_by_name' to 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 433031 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	aef7278af6	res_pjsip: Allow configuration of endpoint identifier query order This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ ........ Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	259e833e88	res_pjsip: Add reason comment. ........ Merged revisions 433005 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e89f83b3ad	main/frame: Don't report empty disallow values as an error In realtime, it is normal to have a database with both 'allow' and 'disallow' columns in the schema. It is perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files, you can't not provide the disallow value. Thus, the empty disallow value causes a spurious WARNING message, which is kind of annoying. This patch makes it so that a 'disallow' value with no ... value ... is ignored. Granted, you can still screw this up as well, as technically specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no 'allow' value in your database. But really, why would you do that? WHY? ASTERISK-16779 #close Reported by: Atis Lezdins ........ Merged revisions 432970 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432971 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	0d52907d2b	func_curl: Don't hold exclusive lock when performing HTTP request. This code originally kept a lock held when performing the HTTP request to ensure that the options provided to curl remain valid. This doesn't seem to be necessary these days and holding the lock caused requests to happen sequentially instead of in parallel. ASTERISK-18708 #close Reported by: Dave Cabot ........ Merged revisions 432948 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432949 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ac1214d9d4	apps/app_sms: Add an option to prevent SMS content from being logged In some countries, privacy laws specify that SMS content cannot be saved by a provider. This patch adds a new option to the SMS application, 'n', which prevents the SMS content from being written to the SMS log. ASTERISK-22591 #close Reported by: Jan Juergens patches: DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	b3fa35786f	core: Fix tab completion of "core set debug channel" CLI command. The "core set debug channel" CLI command mistakenly had source filenames added to its tab completion. This occurred because the CLI generator fell back to the "core set debug" command which permits setting debug at a source filename level. ASTERISK-21038 #close Reported by: Richard Kenner ........ Merged revisions 432944 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432945 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	b4cc056067	FILE: fix retrieval of file contents when offset is specified The loop that reads in a file was not correctly using the offset when determining what bytes to append to the output. This patch corrects the logic such that the correct portion of the file is extracted when an offset is specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by Di-Shi Sun (License 5076) ........ Merged revisions 432935 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432938 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	dc752f515b	apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation This patch corrects the documentation for the AMD application. Specifically: * It documents the maximum_word_length option, which limits the maximum allowed length of a single utterance. * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was undocumented. Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. ASTERISK-19470 #close Reported by: Frank DiGennaro ........ Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432920 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	c52adca396	chan_pjsip: AMI action PJSIPShowEndpoint closes AMI connection on error. Also fixed similar problem with AMI action PJSIPShowEndpoints. ASTERISK-24872 #close Reported by: Dmitriy Serov Review: https://reviewboard.asterisk.org/r/4487/ ........ Merged revisions 432894 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	636d82f4d8	chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	d42c6adb1a	Revert - res_pjsip: Allow configuration of endpoint identifier query order Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	f2c21ead1f	Logger: Fix MALLOC_DEBUG build error. Revision 432834 introduced a build error when MALLOC_DEBUG is used. Switch callid threadstorage to simple AST_THREADSTORAGE since we no longer need custom cleanup. Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	38ee441ea7	main/audiohook: Update internal sample rate on reads When an audiohook is created (which is used by the various Spy applications and Snoop channel in Asterisk 13+), it initially is given a sample rate of 8kHz. It is expected, however, that this rate may change based on the media that passes through the audiohook. However, the read/write operations on the audiohook behave very differently. When a frame is written to the audiohook, the format of the frame is checked against the internal sample rate. If the rate of the format does not match the internal sample rate, the internal sample rate is updated and a new SLIN format is chosen based on that sample rate. This works just fine. When a frame is read, however, we do something quite different. If the format rate matches the internal sample rate, all is fine. However, if the rates don't match, the audiohook attempts to "fix up" the number of samples that were requested. This can result in some seriously large number of samples being requested from the read/write factories. Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)). However, if the audiohook is still expecting an internal sample rate of 8000, we'll attempt to "fix up" the requested samples to: samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate); which is: 92160 = 3840 * (192000 / 8000) This results in us attempting to read 92160 samples from our factories, as opposed to the 3840 that we actually wanted. On a 64-bit machine, this miraculously survives - despite allocating up to two buffers of length 92160 on the stack. The 32-bit machines aren't quite so lucky. Even in the case where this works, we will either (a) get way more samples than we wanted; or (b) get about 3840 samples, assuming the timing is pretty good on the machine. Either way, the calculation being performed is wrong, based on the API users expectations. My first inclination was to allocate the buffers on the heap. As it is, however, there's at least two drawbacks with doing this: (1) It's a bit complicated, as the size of the buffers may change during the lifetime of the audiohook (ew). (2) The stack is faster (yay); the heap is slower (boo). Since our calculation is flat out wrong in the first place, this patch fixes this issue by instead updating the internal sample rate based on the format passed into the read operation. This causes us to read the correct number of samples, and has the added benefit of setting the audihook with the right SLIN format. Note that this issue was caught by the Asterisk Test Suite as a result of r432195 in the 13 branch. Because this issue is also theoretically possible in Asterisk 11, the change is being made here as well. Review: https://reviewboard.asterisk.org/r/4475/ ........ Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432811 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	29304d10a0	Add support for the clang compiler; update RAII_VAR to use BlocksRuntime RAII_VAR, which is used extensively in Asterisk to manage reference counted resources, uses a GCC extension to automatically invoke a cleanup function when a variable loses scope. While this functionality is incredibly useful and has prevented a large number of memory leaks, it also prevents Asterisk from being compiled with clang. This patch updates the RAII_VAR macro such that it can be compiled with clang. It makes use of the BlocksRuntime, which allows for a closure to be created that performs the actual cleanup. Note that this does not attempt to address the numerous warnings that the clang compiler catches in Asterisk. Much thanks for this patch goes to: * The folks on StackOverflow who asked this question and Leushenko for providing the answer that formed the basis of this code: http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang * Diederik de Groot, who has been extremely patient in working on getting this patch into Asterisk. Review: https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133 ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by: Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik de Groot (License 6600) ........ Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432808 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4115e327ac	res_pjsip: Move internal init/destroy prototypes to private header file. Done as a separate commit from a finding in https://reviewboard.asterisk.org/r/4467/ ........ Merged revisions 432787 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	89b65f5dda	res_pjsip: Fix pjsip.conf type=global object default value handling. When a type=global section is not defined in pjsip.conf the global defaults are not applied. As a result the mandatory Max-Forwards header is not added to SIP messages for res_pjsip/chan_pjsip. The handling of pjsip.conf type=global objects has several problems: 1) If the global object is missing the defaults are not applied. 2) If the global object is missing the default_outbound_endpoint's default value is not returned by ast_sip_global_default_outbound_endpoint(). 3) Defines are needed so default values only need to be changed in one place. * Added a sorcery instance observer callback to check if there were any type=global sections loaded. If there were more than one then issue an error message. If there were none then apply the global defaults. * Fixed ast_sip_global_default_outbound_endpoint() to return the documented default when no type=global object is defined. * Made defines for the global default values. * Increased the default_useragent[] size because SVN version strings can get lengthy and 128 characters may not be enough. * Fixed an off-nominal code path ref leak in global_alloc() if the string fields fail to initialize. * Eliminated RAII_VAR in get_global_cfg() and ast_sip_global_default_outbound_endpoint(). ASTERISK-24807 #close Reported by: Anatoli Review: https://reviewboard.asterisk.org/r/4467/ ........ Merged revisions 432766 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	185d2e082a	res_pjsip: Fixed invalid empty Server and User-Agent SIP headers. Setting pjsip.conf useragent to an empty string results in an empty SIP header being sent. * Made not add an empty SIP header item to the global SIP headers list. Review: https://reviewboard.asterisk.org/r/4467/ ........ Merged revisions 432764 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	2889f074a0	core: Don't create snapshots with locks. Snapshots are immutable and are never changed. Allocating them with a lock is wasteful. Review: https://reviewboard.asterisk.org/r/4469/ ........ Merged revisions 432742 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	15d266bf85	res/res_config_odbc: Fix improper escaping of backslashes with MySQL When escaping backslashes with MySQL, the proper way to escape the characters in a LIKE clause is to escape the '\' four times, i.e., '\\\\'. To quote the MySQL manual: "Because MySQL uses C escape syntax in strings (for example, “\n” to represent a newline character), you must double any “\” that you use in LIKE strings. For example, to search for “\n”, specify it as “\\n”. To search for “\”, specify it as “\\\\”; this is because the backslashes are stripped once by the parser and again when the pattern match is made, leaving a single backslash to be matched against." ASTERISK-24808 #close Reported by: Javier Acosta patches: res_config_odbc.diff uploaded by Javier Acosta (License 6690) ........ Merged revisions 432720 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432721 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ab6e2c93f3	app_voicemail: Fix crash with IMAP backends when greetings aren't present When an IMAP backend is in use and greetings are set to be used, but aren't present for a user in their IMAP folder, Asterisk will crash. This occurs due to the mailstream being set to the 'greetings' folder and being left in that particular state, regardless of the success/failure of the attempt to access the folder the mailstream points to. Later access of the mailstream assumes that it points to the 'INBOX' (or some other folder), resulting in either a crash (if the greetings folder didn't exist and the mailstream is invalid) or an inability to read messages from the 'INBOX' folder. This patch restores the mailstream to its correct state after accessing the greetings. This fixes the crash, and sets the mailstream to the state that VoiceMailMain expects. Note that while ASTERISK-23390 also contained a patch for this issue, the patch on ASTERISK-24786 is the one being merged here. Review: https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close Reported by: Ben Smithurst ASTERISK-24786 #close Reported by: Graham Barnett Tested by: Graham Barnett patches: app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685) ........ Merged revisions 432695 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432696 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	79e9b37ad0	localtime: Fix file descriptor leak on kqueue(2) systems The localtime management in the Asterisk core contains a thread that watches for changes in the local timezone. On systems where the directory containing /etc/localtime is modified frequently, the thread monitoring the changes will be woken up to determine if any changes in timezone have occurred. When using kqueue(2), this can cause a leak of file descriptors due to some improper management of resources. This patch updates the kqueue(2) handling in localtime, such that is no longer leaks resources. Review: https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License 6680) ........ Merged revisions 432691 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432693 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e7ee83ea90	res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER. A race condition happened between initiating a transfer and requesting that a dialog termination be delayed. Occasionally, the transferrer channels would exit the bridge and hangup before the dialog termination delay was requested. * Made request dialog termination delay before initiating the transfer action. If the transfer fails then cancel the delayed dialog termination request. ASTERISK-24755 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4460/ ........ Merged revisions 432668 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	1ce529d30e	res_pjsip: allow configuration of endpoint identifier query order It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ ........ Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	a5f80f1781	res_rtp_asterisk: Fix wrongful use of USE_PJPROJECT define. As pjproject is now used as a shared library a different define, HAVE_PJPROJECT, is used to specify if pjproject is present. ASTERISK-24830 #close Reported by: Stefan Engström ........ Merged revisions 432614 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	affcf1d766	res_pjsip_refer: Make safely get the context for a blind transfer. Made safely get the TRANSFER_CONTEXT channel value while the channel is locked in refer_incoming_attended_request() and refer_incoming_blind_request(). The pointer returned by pbx_builtin_getvar_helper() is only valid while the channel is locked. ........ Merged revisions 432594 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	090ab1735b	res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock. The lock is unused. ........ Merged revisions 432574 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Jonathan Rose	b85cb7ea1b	app: Add functions to swap voicemail function table for testing purposes ........ Merged revisions 432556 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	c7cc1b3059	chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. The distinctive ring feature interferes with detecting Caller ID and appears to have been broken for years. What happens is if you have a ring-ring cadence as used in the UK you get too many DAHDI events for the distinctive ring pattern array and Caller ID detection is aborted. I think when Zapata/DAHDI added the ring begin event it broke distinctive ring. More events happen than before and the code does no filtering of which event times are recorded in the pattern array. * Made distinctive ring only record the ringt count when the ring ends instead of on just any DAHDI event. Distinctive ring can be ring, ring-ring, ring-ring-ring, or different ring durations for the up to three rings. * Fixed the distinctive ring detection enable (chan_dahdi.conf option usedistinctiveringdetection) to be per port instead of somewhat per port and somewhat global. This has been broken since v1.8. * Fixed using the default distinctive ring context when the detected pattern does not match any configured dringX patterns. The default context did not get set when the previous call was a matched distinctive ring pattern and the current call is not matched. This has been broken since v1.8. * Made distinctive ring have no effect on Caller ID detection when it is disabled. Caller ID detection just monitors for 10 seconds before giving up. * Fixed leak of struct callerid_state memory when a polarity reversal during Caller ID detection causes the incoming call to be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders Review: https://reviewboard.asterisk.org/r/4444/ ........ Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago

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