asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	0618d89499	app_queue: Initialize array holding MixMonitor exec options If the channel variable MONITOR_EXEC is set, app_queue will pass the specified execution parameters to the MixMonitor application when a queue is recorded. If that channel variable is not set, the buffer that holds the escaped value was not being initialized to NULL, and so would be passed to the MixMonitor application with garbage. Hilarity ensued as app_mixmonitor attempted to execute gobeldy-gook. ........ Merged revisions 399681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e7d49d28ea	Fix a performance problem CDRs There is a large performance price currently in the CDR engine. We currently perform two ao2_callback calls on a container that has an entry for every channel in the system. This is done to create matching pairs between channels in a bridge. As such, the portion of the CDR logic that this patch deals with is how we make pairings when a channel enters a mixing bridge. In general, when a channel enters such a bridge, we need to do two things: (1) Figure out if anyone in the bridge can be this channel's Party B. (2) Make pairings with every other channel in the bridge that is not already our Party B. This is a two step process. In the first step, we look through everyone in the bridge and see if they can be our Party B (single_state_process_bridge_enter). If they can - yay! We mark our CDR as having gotten a Party B. If not, we keep searching. If we don't find one, we wait until someone joins who can be our Party B. Step 2 is where we changed the logic (handle_bridge_pairings and bridge_candidate_process). Previously, we would first find candidates - those channels in the bridge with us - from the active_cdrs_by_channel container. Because a channel could be a candidate if it was Party B to an item in the container, the code implemented multiple ao2_container callbacks to get all the candidates. We also had to store them in another container with some other meta information. This was rather complex and costly, particularly if you have 300 Local channels (600 channels!) going at once. Luckily, none of it is needed: when a channel enters a bridge (which is when we're figuring all this stuff out), the bridge snapshot tells us the unique IDs of everyone already in the bridge. All we need to do is: For all channels in the bridge: If the channel is us or our Party B that we got in step 1, skip it Compare us and the candidate to figure out who is Party A (based on some specific rules) If we are Party A: Make a new CDR for us, append it to our chain, and set the candidate as Party B If they are Party A: If they don't have a Party B: Make a new CDR for them, append us to their chain, and us as Party B Otherwise: Copy us over as Party B on their existing CDR. This patch does that. Because we now use channel unique IDs to find the candidates during bridging, active_cdrs_by_channel now looks up things using uniqueid instead of channel name. This makes the more complex code simpler; it does, however, have the drawback that dialplan applications and functions will be slightly slower as they have to iterate through the container looking for the CDR by name. That's a small price to pay however as the bridging code will be called a lot more often. This patch also does two other minor changes: (1) It reduces the container size of the channels in a bridge snapshot to 1. In order to be predictable for multi-party bridges, the order of the channels in the container must be stable; that is, it must always devolve to a linked list. (2) CDRs and the multi-party test was updated to show the relationship between two dialed channels. You still want to know if they talked - previously, dialed channels were always ignored, which is wrong when they have managed to get a Party B. (closes issue ASTERISK-22488) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/2861/ ........ Merged revisions 399666 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	85d6db6cbe	Fix crash in res_pjsip on load if error occurs, and prevent unloading of res_pjsip and res_pjsip_session. During load time in res_pjsip if an error occurred the operation would attempt to rollback all operations done during load. This is not permitted by PJSIP as it will assert if the operation has not been done. This fix changes the code so it will only rollback what has been initialized already. Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to limitations within PJSIP itself. The library environment can only be changed to a certain extent and does not provide the ability, currently, to deinitialize certain required functionality. (closes issue ASTERISK-22474) Reported by: Corey Farrell ........ Merged revisions 399624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ec5a724714	res_rtp_asterisk: Fix ref leaks in ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the loop so it is unref'ed after every loop. Moved message_blob to loop and switched it to a regular variable. The regular variable was used since message_blob is used in a very contained way. (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches: rtcp_report-leak.patch (license #5909) patch uploaded by Corey Farrell Tested by: Corey Farrell ........ Merged revisions 399607 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	46da169b6d	media_index: Fix process_description_file() memory leak of file_id_persist. ........ Merged revisions 399596 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	dbec6e92d1	features_config: Fix config ref leak of parkinglots. This leak happend for just about every channel created. ........ Merged revisions 399585 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5afbc01d5d	app_queue: Fix json blob ref leak. The json ref from queue_member_blob_create() was never released. ........ Merged revisions 399583 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	120abb5ecd	json: Make it obvious that ast_json_unref() is NULL safe. It looked like the safety check was done after the NULL pointer was used. ........ Merged revisions 399576 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6cc3084ae7	Ensure global types in the config framework are initialized If a config object was allocated but one of its global objects was never encountered, then the global object's defaults were never applied. Ensure that global objects are initialized properly upon allocation instead of on configuration. Review: https://reviewboard.asterisk.org/r/2866/ ........ Merged revisions 399564 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	638577bef7	originate/call forwarding: Fix a crash when forwarding a call from originate (closes issue ASTERISK-22487) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/2868/ ........ Merged revisions 399553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d4a026a0ee	Add a missing session supplement unregistration in chan_pjsip for ACKs. (closes issue ASTERISK-22453) Reported by: Corey Farrell Patches: chan_pjsip_session_unregister_supplement.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 399531 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2d091df520	Fix memory leak in logger. Fixed a memory leak discovered in the logger where a temporary string buffer was not being freed. (closes issue ASTERISK-22540) Reported by: John Hardin ........ Merged revisions 399513 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399514 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	cf9272c05c	optional_api: Make always use the standard malloc functions even with MALLOC_DEBUG. ........ Merged revisions 399501 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e89e19c479	chan_sip: Make direct media reinvites for T38 put Asterisk in the media path Prior to this patch, Asterisk would incorrectly use the previous endpoint addresses in SDP in spite of providing its own port. T38 is never meant to be done through directmedia and Asterisk should always be in the media path for these streams. (closes issue ASTERISK-17273) Reported by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ ........ Merged revisions 399456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399459 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d5372f34df	Fix jitter buffer log file creation This adjusts '/'-to-'#' replacement to replace all instances of '/' instead of just the first to ensure that the jitter buffer log file gets the correct name as per Richard Kenner's suggestion. (closes issue ASTERISK-21036) Reported by: Richard Kenner ........ Merged revisions 399402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399403 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399404 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d393812203	Update prep_tarball with new documentation files on the Asterisk wiki This will now pull both a command reference for the version being prepared, as well as an Admin Guide that applies to all versions of Asterisk. (issue ASTERISK-22439) Reported by: Olle Johansson ........ Merged revisions 399351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399373 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399376 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	656843dd34	Add a WARNING in bridge_softmix when a timing module isn't loaded If bridge_softmix fails to be created because no timing source is present in Asterisk, this will currently fail gracefully but with (most likely) a generic error message by whatever module tried to create the softmix bridge. This patch adds a more explicit warning so you can actually diagnose and fix the problem. Review: https://reviewboard.asterisk.org/r/2857/ ........ Merged revisions 399353 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399365 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	28da6dc0a5	Make config framework able to reload module configs with multiple config files. The config framework is supposed to be able to load configs that come from multiple config files. The principle example is chan_sip's sip.conf and users.conf. Unfortunately, it only does this correctly on initial load. This patch causes the module's config to be reloaded entirely if any of the config files change. (closes issue ASTERISK-22009) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/2859/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	0e1d364fbd	res_pjsip_messaging: Register message technology as pjsip pjsip's message technology was being registered as 'sip', which was causing it to not load due it conflicting with chan_sip's registered 'sip' technology for messaging. It now registers as 'pjsip'. However, due to this change the "to" field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to res_pjsip_messaging will automatically have their "to" fields altered in order to accommodate the change. Outgoing messages also handle changing it back to 'sip' before being sent so the pjsip library will properly handle it. (closes issue ASTERISK-22445) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2833/ ........ Merged revisions 399339 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	38fa628812	Fix Segfault In features-config.c When Application Has No Arguments Some applications do not require arguments. Therefore, when parsing application maps in features.conf, it is possible that app_data will be set to NULL. * This patch sets app_data to "" if it is NULL. Review: https://reviewboard.asterisk.org/r/2804 ........ Merged revisions 399294 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	391f0003c4	Change the "external_media_address" PJSIP endpoint option to "media_address". The endpoint option does not apply to communication with external entities. Rather, the option is applied to all communications with the endpoint. The external_media_address transport configuration option may override the endpoint option if it turns out that we are going to be communicating with an external entity. Two things of note: 1) I have not updated the XML documentation. This is being taken care of by Rusty as part of his work on issue ASTERISK-22405 2) This commit is likely to cause testsuite failures since there are tests that use the external_media_address endpoint option, and they will need to be changed over. Well, I'm planning to get that updated ASAP after this commit. (closes issue ASTERISK-22528) reported by Rusty Newton ........ Merged revisions 399283 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	667fa56b1b	Remote console: more output discrepancies The remote console continued to have issues with its output. In this case CLI command output would either not show up (if verbose level = 0) or would contain verbose prefixes (if verbose level > 0) once log messages were sent to the remote console. The fix now now adds verbose prefix data to all new lines contained in a verbose log string. (closes issue ASTERISK-22450) Reported by: David Brillert (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2825/ ........ Merged revisions 399267 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399268 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e6e73cbc45	Fix doxygen to use correct units of features.conf options. ........ Merged revisions 399257 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	375c2f5a5c	Fix other timeouts (atxferloopdelay and atxfernoanswertimeout) to use seconds instead of milliseconds. Thanks to Richard Mudgett for pointing this out. ........ Merged revisions 399247 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f653bfa1f3	Switch transferdigittimeout to be configured as seconds instead of milliseconds. This was an unintentional consequence of the update of features.conf to use the config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk developers list for pointing out the problem. ........ Merged revisions 399237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	b1db2df871	Confbridge: empty conference not being torn down Confbridge would not properly tear down an empty conference bridge when all users were kicked via end_marked=yes and at least one user was also set to wait_marked. This occurred because while end_marked users were being kicked and at least one was also set to wait_marked then the leave wait_marked handler would be called on that user, but there would be no waiting user (still considered active). The waiting users would decrement and now be negative. The conference would remain, but be put into an inactive state. The solution was to move from the active list to the wait list, those users with wait_marked set right before kicking. This allows both the active and wait users to decrement correctly and the confbridge to tear down properly. A crashed also occurred when trying to list the specific conference from the CLI. This happened because the conference specified was invalid. Since the conference properly tears down now there is no way to reference it thus alleviating the crash as well. (closes issue ASTERISK-21859) Reported by: Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/ ........ Merged revisions 399222 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399225 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e77fba4b25	Fix module load errors for test_ari_model.so. You cannot use a function pointer variable with an external function from another dynamically loaded module because data variables are always resolved even with RTLD_LAZY. * Added wrapper functions for ast_ari_validate_int() and ast_ari_validate_string() to use instead for the function pointer variable. (closes issue ASTERISK-22457) Reported by: David M. Lee ........ Merged revisions 399207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	10d4ed93ff	app_speech_utils: Fix unresolved symbol ast_speech_get_setting(). Fixes regression introduced by -r374096. * Made res_speech.export.in export ast_* symbols instead of specific functions. * Made app_speech_utils.c declare that it is dependent upon res_speech. (issue ASTERISK-17136) Reported by: Richard Kenner ........ Merged revisions 399197 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	819359dcfd	chan_iax2: Fix saving the wrong expiry time in astdb. When a new IAX2 client registers, the astdb database is updated with the value of minregexpire defined in iax.conf instead of using the expiry time that is provided by the client. The provided expiry time of the client is updated after inserting the astdb entry. As a consequence, restarting or reloading asterisk creates clients whose registration may expire before they reregister. The clients are therefore unavailable after minregexpire seconds until they reregister. * Move updating of the expiry time to before inserting into the astdb. (closes issue ASTERISK-22504) Reported by: Stefan Wachtler Patches: chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler ........ Merged revisions 399158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399159 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399160 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	376d277b02	Filter internal channels out of bridge enter/leave message handling Some channels exist merely as an implementation detail in Asterisk, such as ConfBridge's announcer/recorder channels. These channels should never be exposed to the outside world, or to interfaces that report on Asterisk. We already filter out such channels in snapshot processing; however, we failed to filter out bridge related messages that involved these channels. This patch filters out bridge related messages that are for such channels. This prevents a spurious WARNING message from being displayed when those channels move in and out of bridges. ........ Merged revisions 399146 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2a371cd80b	Restore Dial, Queue, and FollowMe 'I' option support. The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	03c7857375	Don't write to /tmp/refs when REF_DEBUG is not defined. If MALLOC_DEBUG is enabled, then the debug destructor for the container is used, which would erroneously write to /tmp/refs. This patch only uses the debug destructor if ref_debug is used. (closes issue ASTERISK-22536) ........ Merged revisions 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399099 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399100 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	9deb416397	Create more accurate Contact headers for dialogs when we are the UAS. (closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	bbf5fbbd8c	Change how realms are handled for outbound authentication. With this change, if no realm is specified in an outbound auth section, then we will simply match the realm that was present in the 401/407 challenge. (closes issue ASTERISK-22471) Reported by George Joseph (closes issue ASTERISK-22386) Reported by Rusty Newton Patches: outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322) ........ Merged revisions 399059 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	8c24c69724	Recorded merge of revisions 399035,399049 from http://svn.asterisk.org/svn/asterisk/branches/12 These were lost in r399071 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	0ab5d3015d	Put merge tracking for r399039 back. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	873969d6c5	Broke the build! Forgot para tags within my description. https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 ........ Merged revisions 399064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2a57f6ccf7	res_pjsip: Forward PJSIP logging to Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log messages to Asterisk's logger. This is done in a new module: res_pjsip_log_forwarder.so. This patch sets defaultenabled on the existing res_pjsip_logger.so to no, since logging every SIP packet seems a bit odd to do by default, and is (hopefully) less necessary with regular PJSIP logging. It also removes res_rtp_asterisk's disabling of PJSIP logging. (closes issue ASTERISK-22360) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2830/ ........ Merged revisions 399049 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	f56796a539	ARI: Fix WebSocket response when subprotocol isn't specified When I moved the ARI WebSocket from /ws to /ari/events, I added code to allow a WebSocket to connect without specifying the subprotocol if there's only one subprotocol handler registered for the WebSocket. Naively, I coded it to always respond with the subprotocol in use. Unfortunately, according to RFC 6455, if the server's response includes a subprotocol header field that "indicates the use of a subprotocol that was not present in the client's handshake [...], the client MUST _Fail the WebSocket Connection_.", emphasis theirs. This patch correctly omits the Sec-WebSocket-Protocol if one is not specified by the client. (closes issue ASTERISK-22441) Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0ffcd11380	Fix several crashes in MeetMeAdmin This change ensures that MeetMeAdmin commands requiring a user actually get a user and fixes another issue where an extra dereference could occur for a last-entered user being ejected if a user identifier was also provided. (closes issue ASTERISK-21907) Reported by: Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/ ........ Merged revisions 399033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399034 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399035 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	fc09e5eb66	'identify' configObject doesn't have a synopsis Add a straightforward synopsis and description to the identify config object in XML documentation. (issue ASTERISK-22311) (closes issue ASTERISK-22311) Reported By: Rusty Newton ........ Merged revisions 399031 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ef53242700	CLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab completion. These two commands must deal with the live bridges container for tab completion and not the stasis cache. ........ Merged revisions 399021 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	94754227a6	astobj2: Register the bridges container for debug inspection. ........ Merged revisions 399019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	1b777d8946	Documentation fix and improvements to XML configuration help res_pjsip_acl * One bug fix. Made the synopsis for "type" to accurate. * changing the usage of "IP-domains" to "IP addresses" * clarifying the usage for the options, by adding a relevant description for each * modified other areas of the XML help for clarity, such as the module description and a few synopsis changes here and there. See the patch. (issue ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty Newton Review: https://reviewboard.asterisk.org/r/2823/ ........ Merged revisions 399017 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	039030f245	chan_sip: Revert r398835 due to failing tests involving originate (issue ASTERISK-22424) Reported by: Jonathan Rose ........ Merged revisions 398977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398986 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398991 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5be186d7c5	core_local: Fix memory corruption race condition. The masquerade super test is failing on v12 with high fence violations and crashing. The fence violations are showing that party id allocated memory strings are somehow getting corrupted in the bridge_reconfigured_connected_line_update() function. The invalid string values happen to be the freed memory fill pattern. After much puzzling, I deduced that the bridge_reconfigured_connected_line_update() is copying a string out of the source channel's caller party id struct just as another thread is updating it with a new value. The copying thread is using the old string pointer being freed by the updating thread. A search of the code found the unreal_colp_redirect_indicate() routine updating the caller party id's without holding the channel lock. A latent bug in v1.8 and v11 hatched in v12 because of the bridging and connected line changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2839/ ........ Merged revisions 398938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	6ad74509f3	Fix symbol collision with pjsua. We shouldn't be exporting any symbols that start with pjsip_. ........ Merged revisions 398927 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	4e3f78ad7b	'queue add member' help text correction You are adding dial strings to the queue, not channels. An aribitrary string could be used, but you are typically referencing a channel. Correcting the command help text. (issue ASTERISK-22263) (closes issue ASTERISK-22263) Reported By: Rusty Newton ........ Merged revisions 398884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398885 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398886 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	7c346a31ef	Documentation fix - waitfordialtone is not boolean, it's time in milliseconds Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport ........ Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398881 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	187802eeb2	chan_sip: Reject calls without prior SDP on 200 OK If we receive a 200 OK without SDP, we will now check to see if the remote address has been established for that channel's RTP session and if the to tag for that channel has changed from the most recent to tag in a response less than 200. If either a change has been made since the last to-tag was received or the remote address is unset, then we will drop the call. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header ........ Merged revisions 398835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398836 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398837 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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