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Author	SHA1	Message	Date
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
Richard Mudgett	f1ab2c5e8b	chan_sip: Fix realtime locking inversion when poking a just built peer. ... When a realtime peer is built it can cause a locking inversion when the just built peer is poked. If the CLI command "sip show channels" is periodically executed then a deadlock can happen because of the locking inversion. * Push the peer poke off onto the scheduler thread to avoid the locking inversion of the just built realtime peer. AST-1540 ASTERISK-24838 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4454/ ........ Merged revisions 432526 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432528 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	5c3e33b3ca	app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE. ... There is a leftover "assert" in app_voicemail/__messagecount that references variables that don't exist. This causes the compile to fail when --enable-dev-mode and IMAP_STORAGE are selected. This patch removes the assert. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4461/ ........ Merged revisions 432484 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432485 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	41ba8fd7c0	translate: Prevent invalid memory accesses on fast shutdown ... When a 'core restart now' or 'core stop now' is executed and a channel is currently in a media operation, the translator matrix can be destroyed while a channel is currently blocked on getting the best translation choice (see ast_translator_best_choice). When the channel gets the mutex, the translation matrix now has invalid memory, and Asterisk crashes. This patch does two things: (1) We now only clean up the translation matrix on a graceful shutdown. In that case, there are no channels, and so there is no risk of this occurring. (2) We also now set the __matrix and __indextable to NULL. In some initial backtraces when this occurred, it looked as if there was a memory corruption occurring, and it wasn't until we determined that something had restarted Asterisk that the issue became clear. By setting these to NULL on shutdown, it becomes a bit easier to determine why a crash is occurring. Note that we could litter the code with NULL checks on the __matrix, but the act of making the translation matrix cleaned up on shutdown should preclude this issue from occurring in the first place, and this part of the code needs to be as fast as possible. Review: https://reviewboard.asterisk.org/r/4457/ ........ Merged revisions 432453 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	278ea2f468	res/res_pjsip_sdp_rtp: Revert portion of r432195 ... Unfortunately, while initial testing with ConfBridge did not reproduce the audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing did show that bridge_softmix and/or ConfBridge has a severe problem bridging two or more participants at different sampling rates. Sometimes, it even picks odd sampling rates that cause hideous audio problems. This patch backs out the offending portion of the code until the issues in the affected bridging modules can be more properly analyzed. ASTERISK-24841 ........ Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	9e841e4fb6	ARI: Fix crash if integer values used in JSON payload 'variables' object. ... Sending the following ARI commands caused Asterisk to crash if the JSON body 'variables' object passes values of types other than strings. POST /ari/channels POST /ari/channels/{channelid} PUT /ari/endpoints/sendMessage PUT /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(), ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751 #close Reported by: jeffrey putnam Review: https://reviewboard.asterisk.org/r/4447/ ........ Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	d79670b269	Dial API: add self destruct option when complete ... This patch adds a self-destruction option to the dial api. The usefulness of this is mostly when using async mode to spawn a separate thread used to handle the new call, while the calling thread is allowed to go on about other business. The only alternative to this option would be the calling thread spawning a new thread, or hanging around itself waiting to destroy the dial struct after completion. Example of use (minus error checking): struct ast_dial *dial = ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo"); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1); The dial_run call will return almost immediately after spawning the new thread to run and monitor the dial. If the call is answered, it is placed into the echo app. When completed, it will call ast_dial_destroy() on the dial structure. Note that any allocations made to pass values to ast_dial_set_user_data() or dial options must be free'd in a state callback function on any of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED, AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review: https://reviewboard.asterisk.org/r/4443/ ........ Merged revisions 432385 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	d04fbb0f9d	app_chanspy, channel: fix frame leaks ... Fixed a couple of frame leaks that were found during testing. ASTERISK-24828 #close Reported by: John Hardin Review: https://reviewboard.asterisk.org/r/4445/ ........ Merged revisions 432362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432363 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	8a16c2f0c2	make: Remove 'res_features' from libraries to link against with cygwin/mingw32 ... Both the apps and channels Makefiles still listed 'res_features' as modules to link against when compiling for cygwin or mingw32. This module hasn't existed for quite some time. ASTERISK-18105 #close Reported by: feyfre ........ Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432342 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	3725173b9e	channels/chan_sip: Don't send a BYE after final response when PBX thread fails ... When Asterisk fails to start a PBX thread for a new channel - for example, when the maxcalls setting in asterisk.conf is exceeded - we currently send a final response, and then attempt to send a BYE request to the UA. Since that's all sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt such that we don't get stuck sending BYE requests to something that does not want it. Note that this patch is a slight modification of the one on ASTERISK-15434. For clarity, it explicitly calls sipalreadygone with the calls to transmit a final response. ASTERISK-21845 ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027) ........ Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Rusty Newton	e484140aed	configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1 ... Example configuration files for a "basic PBX" deployment for the fictitious Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/ and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company Reported by: Malcolm Davenport Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/4379/ ........ Merged revisions 432301 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ced84d7e62	configure: Promote SQLite3 "not installed" warning to error ... Since Asterisk won't build without the library, not having it is definitely an error. Thanks to Kyle Kurz for pointing this out. ........ Merged revisions 432280 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432281 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	4b63da7f7d	channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario ... When we receive an SDP as part of an offer/answer for a peer/friend has been configured to require encryption, and that SDP offer/answer failed to provide acceptable crypto attributes, we currently issue a WARNING that uses the phrase "we" and "requested". In this case, both of those terms are ambiguous - the user will probably think "we" is Asterisk (it most likely isn't) and it may not be a "request", so much as an SDP that was received in some fashion. This patch makes the WARNING messages slightly less bad and a bit more accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton ........ Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432278 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	d68012d1a3	channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI ... Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would be rejected if those crypto attributes contained either a key lifetime or a MKI parameter. While from a theoretical point of view this was defensible - Asterisk does not support key lifetimes or multiple crypto keys - from a practical point of view, this is quite a problem. A large number of endpoints offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually have to support anything more than a single key or refresh the key. In reality, this is (so far as we've seen) always the case. This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters in the following fashion: > The Lingon branch now handle lifetime and MKI parameters. > > We only accept lifetimes up to max for the crypto and higher than 10 hours > for packetization of 20 ms (50 pps). > > We only handle MKI with index 1. > > We do not really bother with counting packets and reinviting at end of > lifetime, so the min of 10 hours kind of takes care of most calls. If there > are longer ones, we rely on the other side for re-invites. > > It's still not perfect, but I personally think this is an improvement. A > configuration option for minimum lifetime accepted could be added. When the patch was ported forward, I decided against adding a configuration option as Olle's handling was more than sufficient for every case I've seen come through the issue tracker or through interoperability testing. We can revisit that decision if it proves to be false. A few small other tweaks were made to the surrounding code to reduce indentation and provide better type safety for the 'tag' parameter. Review: https://reviewboard.asterisk.org/r/4419/ Review: https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close Reported by: Terry Wilson ASTERISK-17899 #close Reported by: Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432258 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
David M. Lee	ff642289f4	Increase WebSocket frame size and improve large read handling ... Some WebSocket applications, like [chan_respoke][], require a larger frame size than the default 8k; this patch bumps the default to 16k. This patch also fixes some problems exacerbated by large frames. The sanity counter was decremented on every fread attempt in ws_safe_read(), regardless of whether data was read from the socket or not. For large frames, this could result in loss of sanity prior to reading the entire frame. (16k frame / 1448 bytes per segment = 12 segments). This patch changes the sanity counter so that it only decrements when fread() doesn't read any bytes. This more closely matches the original intention of ws_safe_read(), given that the error message is "Websocket seems unresponsive". This patch also properly logs EOF conditions, so disconnects are no longer confused with unresponsive connections. [chan_respoke]: https://github.com/respoke/chan_respoke Review: https://reviewboard.asterisk.org/r/4431/ ........ Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432237 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	57525c3cf2	config.h: Use real parameter names for ast_variable_new() define. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	8574c4d197	channels/chan_sip: Fix crash when transmitting packet after thread shutdown ... When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE ........ Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	a528dfc9a7	ARI/PJSIP: Apply requesting channel's format cap to created channels ... This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan ........ Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	91733b5d15	bridge_softmix: G.729 codec license held ... When more than one call using the same codec type enters into a softmix bridge and no audio is present for a channel the bridge optimizes the out frame by using the same one for all channels with the same codec type. Unfortunately, when that number (channels with same codec type) dropped to <= 1 the codec was not dereferenced. At least not until all parties left the bridge. Thus in the case of G.729 the license was not released. This patch ensures that the codec is dereferenced immediately when the optimization no longer applies. ASTERISK-24797 #close Reported by: Luke Hulsey Review: https://reviewboard.asterisk.org/r/4429/ ........ Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432175 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	bedf51b2ce	res_ari_channels: Return a 404 response when a requested channel variable does not exist. ... This change makes it so that if a channel variable is requested and it does not exist a 404 response will be returned instead of an allocation failed response. This makes it easier to debug and figure out what is going on for a user. ASTERISK-24677 #close Reported by: Joshua Colp ........ Merged revisions 432154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	87b7060f36	res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER. ... Some implementations don't pay attention to the expires for individual contacts. In this case they may consider the lack of an Expires header in the 200 OK as unregistered. This change makes it so if an Expires header is present in the REGISTER we will add one in the 200 OK. ASTERISK-24785 #close Reported by: Ross Beer ........ Merged revisions 432136 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	283bb15c16	res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid. ... ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged revisions 432118 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	b3c1ad5d73	apps/app_voicemail: Demote an ERROR message to a WARNING message ... When using IMAP voicemail with FreePBX, you will often get ERROR messages complaining about not being able to find a mailbox. This is due to how FreePBX handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this a configuration error, as in any other system it would be indicative of someone misconfiguring their system. Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch demotes the message so that system administrators can hopefully reduce some of the noise in their log files. Note that in the original patch this was made into a NOTICE, but that's a too forgiving. ASTERISK-24790 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) ........ Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432099 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	2ea7ccbf70	http: Add missing html tag to 'httpstatus' functionality. ... ASTERISK-24724 #close Reported by: Ashley Sanders ........ Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	e66b874f5d	Allow shutdown to unload modules that register bucket scheme's or codec's. ... * Change __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules that call ast_bucket_scheme_register or ast_codec_register to be unloaded during graceful shutdown only (13+ only). ASTERISK-24796 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4428/ ........ Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432059 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	bb71672a47	main/asterisk.c: Reverse #if statement in listener() to fix code folding. ... listener() opens the same code block in two places (#if and #else). This confuses some folding editors causing it to think that an extra code block was opened. Folding in 'geany' causes all code after listener() to be folded as if it were part of that procedure. ASTERISK-24813 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4435/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	ce50fa314a	asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers. ... Add a couple of missing closing brackets / parenthesis. ASTERISK-24814 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4436/ ........ Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432055 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	bb06603d5f	chan_dahdi/sig_analog: Put log message strings on one line. ... With the log messages on one line, you can search for the log message seen in the log and expect to find it. ........ Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432034 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	340818ad12	ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk. ... Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from realtime. Turns out it was just missing a call ast_sorcery_apply_config(). res_pjsip_acl was missing it as well, so I added it. The other pjsip modules looked OK. ASTERISK-24811 #close Reported-by: Matt Hoskins Tested-by: George Joseph Tested-by: Matt Hoskins patches: res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688) Review: https://reviewboard.asterisk.org/r/4433/ ........ Merged revisions 432033 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago