asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	b8fa8aa775	clang compiler warnings: Fix pointer-bool-converesion warnings This patch fixes several warnings pointed out by the clang compiler. * chan_pjsip: Removed check for data->text, as it will always be non-NULL. * app_minivm: Fixed evaluation of etemplate->locale, which will always evaluate to 'true'. This patch changes the evaluation to use ast_strlen_zero. * app_queue: - Fixed evaluation of qe->parent->monfmt, which always evaluates to true. Instead, we just check to see if the dereferenced pointer evaluates to true. - Fixed evaluation of mem->state_interface, wrapping it with a call to ast_strlen_zero. * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero. Review: https://reviewboard.asterisk.org/r/4541 ASTERISK-24917 Reported by: dkdegroot patches: rb4541.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	7bc2345fb1	clang compiler warnings: Fix -Wabsolute-value warnings This patch fixes several warnings caught by clang - in this case, usage of the abs function on non-integer values. This patch uses labs and fabs, as appropriate, in the various affected files. Review: https://reviewboard.asterisk.org/r/4525 ASTERISK-24917 Reported by: dkdegroot patches: rb4525.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433750 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	d2776d4d45	clang compiler warnings: Fix a variety of "unused" warnings This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable errors caught by clang. Specifically: * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], qsmp_cmd_usage[] * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" * funcs/func_env.c:729: Fixed ast_str_append_substr. * main/editline/np/strlcat.c: removed unused rcsid variable * main/editline/np/strlcpy.c: removed unused rcsid variable * main/security_events.c: removed unused TIMESTAMP_STR_LEN * utils/conf2ael.c: removed unused cfextension_states * utils/extconf.c: removed unused cfextension_states Review: https://reviewboard.asterisk.org/r/4526 ASTERISK-24917 Reported by: dkdegroot patches: rb4526.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 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
Richard Mudgett	c7ea108e02	Revert -r430452 It needs to be redone for the next major AMI version change instead. ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	ef34a05f21	AMI: Remove no longer used parameter from astman_send_listack(). Follow-up issue to -r430435 from reviewboard review. ASTERISK-24049 Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	52a7cdb101	AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	1106e8fd0f	main/stasis: Allow subscriptions to use a threadpool for message delivery Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2d9471ab1f	Fix race condition that could result in ARI transfer messages not being sent. From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e4374a3abe	app_queue: fix a couple leaks to struct call_queue in set_member_value set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged revisions 426805 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426806 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426807 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2290393273	app_queue: Cleanup ao2_iterator Clean ao2_iterator, resolving reference leak to queue members. ASTERISK-24454 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4111/ ........ Merged revisions 426255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426260 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cc11a78869	app_queue: Add dialplan function to get the channel name at the specified position in a queue. The QUEUE_GET_CHANNEL function returns the caller's channel name at the specified position in a queue. QUEUE_GET_CHANNEL(<queuename>[,<position>]) The queue position parameter defaults to 1 if not specified. Noop(${QUEUE_GET_CHANNEL(queuename, 2)}) "SIP/peer-00000002", if queue exist and have at least 2 callers Noop(${QUEUE_GET_CHANNEL(queuename, 1)}) Noop(${QUEUE_GET_CHANNEL(queuename)}) "SIP/peer-00000000", if queue exist and have at least 1 caller ASTERISK-24365 #close Reported by: Kristian Hogh Patches: queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh rb4035.patch (license #6639) patch uploaded by Kristian Hogh Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL on reviewbord. Review: https://reviewboard.asterisk.org/r/4035/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	add46fd27c	app_queue: Add RealTime support for queue rules This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) ........ Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2ce95d9d2	accountcode: Slightly change accountcode propagation. The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9cc1a8e893	stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	fa8c58fefb	app_queue: delayed state can cause early leavewhenempty ringing In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont ........ Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415835 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415836 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f2ca3438e7	app_queue: Extend documentation for various Manager actions and events. ........ Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413486 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413487 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cbe7f65674	app_dial and app_queue: Make lock the forwarding channel while taking the channel snapshot. * Fixed ast_channel_publish_dial_forward() not locking the forwarded channel when taking the channel snapshot. * Fixed app_dial.c:do_forward() using the wrong channel to get the original call forwarding string. * Removed unnecessary locking when calling ast_channel_publish_dial() and ast_channel_publish_dial_forward() in app_dial and app_queue. Holding channel locks when calling ast_channel_publish_dial_forward() with a forwarded channel could result in pausing the system while the stasis bus completes processsing a forwarded channel subscription. Review: https://reviewboard.asterisk.org/r/3451/ ........ Merged revisions 412579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	5b7a769fd8	(mix)monitor: Add options to enable a periodic beep Add an option to enable a periodic beep to be played into a call if it is being recorded. If enabled, it uses the PERIODIC_HOOK() function internally to play the 'beep' prompt into the call at a specified interval. This option is provided for both Monitor() and MixMonitor(). Review: https://reviewboard.asterisk.org/r/3424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	76d5c4ed43	app_queue: Re-add HoldTime to QueueCallerAbandon event (simple typo during ast12 refactor). Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........ Merged revisions 411811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c7b8633c26	app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ ........ Merged revisions 411584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411585 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411586 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abf1d883f7	app_queue: Fix documented AMI event name During the rewrite of AMI events to use the Stasis bus, the name of the QueueMemberPaused event was changed to QueueMemberPause. This corrects documentation to reflect that. ........ Merged revisions 409234 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	b17c80f4f0	apps/app_queue - Fix incorrect Macro parameter documentation Macro is executed on the called channel, not the calling channel. (closes issue ASTERISK-23069) Reported By: Bryan Anderson ........ Merged revisions 408447 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408448 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408449 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	28c0cb28d0	channel locking: Add locking for channel snapshot creation Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3a5cc054ed	res_stasis: Expose event for call forwarding and follow forwarded channel. This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ ........ Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1212906351	Reverting r403311. It's causing ARI tests to hang. ........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	483d127d55	app_queue: Honor penalty limits of 0 In the current app_queue code from 1.8 up to trunk the upper and lower penalties can be set to 0 but the value is interpreted to be disabled instead of actually setting limits. This is especially evident if min and max limits are set to 0 and members with penalties of 0 and 1 are in the queue since the member with penalty 1 will still receive calls. This patch adjusts the special disabled value to be INT_MAX instead of 0. (closes issue ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/ Reported by: Schmooze Com ........ Merged revisions 402645 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402646 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402647 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	cdfbc02df1	app_queue: crash if first agent is "busy" If the first agent/member (via CLI "queue show") in a queue is "busy" (dnd, circuit busy, etc...) and no agents answered then app_queue would crash. This occurred because while the calling of agent(s) remained valid the channel on "busy" agent would be set to NULL and then later dereferenced upon a second "rna" function call. The original intention of the code is to have only valid "call attempt" objects (channels != NULL) checked while attempting to call agent(s). It does this by building a "call_next" list of valid "call attempt" objects. In the case of the "busy" agent subsequent builds of the valid "call attempt" list would sometimes include (the case mentioned above) an invalid "call attempt" object. The fix was to make sure the "call attempt" list was appropriately built on every iteration. A NULL sanity check was also added at the original offending spot of the crash just in case another one slipped by somehow. (closes issue ASTERISK-22644) Reported by: Marco Signorini Review: https://reviewboard.asterisk.org/r/2983/ ........ Merged revisions 402517 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ae7a4b1ba	app_queue: Fix CLI "queue remove member" queue_log entry. The queue_log entry resulting from CLI "queue remove member" when log_membername_as_agent is enabled is wrong. It always uses the interface name instead of the member name in the queue_log entry. * Get the queue member before removing it from the queue so the member name is available for the queue_log entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve Patches: fix_membername.diff (license #6505) patch uploaded by Oscar Esteve (modified to fix potential ref leak) ........ Merged revisions 401433 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	e694a976f6	Don't check all realtime queues when doing "queue show some_queue". When using realtime queues, queues have to be fetched from the database every now and then to see if any info has been changed or to see if the queue has been removed. When fetching info for an individual queue, the pruning of other queues is unnecessarily costly. Review: https://reviewboard.asterisk.org/r/2907/ ........ Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401076 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401077 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8eec8fbf83	Make app_queue and res_agi independent of AMI being enabled. The https://reviewboard.asterisk.org/r/2888/ review changes manager to not subscribe to stasis when it is disabled for performance reasons. When manager is disabled app_queue and res_agi decline to load and fail to clean up what they have already allocated. * Made app_queue and res_agi clean up allocated resources when they decline to load. * Made app_queue and res_agi use their own subscriptions to the stasis topics instead of borrowing manager's message router structure inappropriately. (closes issue ASTERISK-22604) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2902/ ........ Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	96d27333d2	Miscellaneous stand alone comment cleanups. ........ Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	0e218f76e2	app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields Commit r62462 added two extra fields for logging "the original position the caller entered the queue at, and the amount of time the caller was waiting in the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two fields disappeared. Those two extra fields were not logged in 1.4 and when the patch was merged, those fields went away. Therefore, this is a regression and was caught by the reporter because he was reading the awesome "Asterisk: The Definitive Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M. Tested by: Dalius M. Patches: asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2901/ ........ Merged revisions 400622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400623 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2de42c2a25	Multiple revisions 399887,400138,400178,400180-400181 ........ r399887 \| dlee \| 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) \| 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 \| dlee \| 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) \| 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 \| dlee \| 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) \| 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 \| dlee \| 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) \| 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 \| dlee \| 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) \| 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ccab0f27bc	app_queue: Make manager events tolerant of Local channel shenanigans app_queue currently attempts to handle Local channel optimizations in an effort to provide accurate information in Stasis messages (and their corresponding AMI events) as well as the Queue log. Sometimes, however, things don't go as planned. Consider the following scenario: SIP/foo <-> L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local channel optimization. app_queue will normally do the following: * Listen for the Local optimization events and update our agent accordingly to SIP/agent in the queue log and messages * When we get a hangup, publish the AgentComplete event based on our information (SIP/foo and SIP/agent) However, as with all things that depend on sanity from something as capricious as Local channels, things can go wrong: (1) SIP/agent immediately hangs up upon answering. This triggers a race condition between termination messages coming from SIP/agent and the ongoing Local channel optimization messages. (Note that this can also occur with SIP/foo) (2) In a race condition, Asterisk can (rarely) deliver the hangup messages prior to the Local channel optimization. In that case, the messages may arrive to app_queue in the following order: * Hangup SIP/Agent * Hangup SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When app_queue receives the hangup of the agent or the caller, it will attempt to publish the AgentComplete event. However, it now has a problem - it thinks its agent is the ;1 side of the Local channel, as it never received the optimization event. At the same time, that channel is already gone. This results in getting NULL from the Stasis cache. What's more, we can't really wait for the optimization message, as we are currently handling the hangup of the channel that the optimization event would tell us to use. This patch modifies the behavior in app_queue such that, since we still have a lot of pertinent queue information (interface, queue name, etc.), we now raise the event with what information we know. The channels involved now may or may not be present. Users will still at least get the "AgentComplete" event, which "completes" the known Agent information. Review: https://reviewboard.asterisk.org/r/2878/ (closes issue ASTERISK-22507) Reported by: Richard Mudgett ........ Merged revisions 400060 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	57e652f2ac	app_queue: Don't be quite so aggressive in initializing the array We only need the first character. ........ Merged revisions 399695 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
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
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
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
Mark Michelson	0bc2a77365	Multiple revisions 397921-397922 ........ r397921 \| mmichelson \| 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) \| 6 lines Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events. Attempting to transfer an unbridged call would result in crashes in either CEL code or in the conversion to AMI messages. ........ r397922 \| mmichelson \| 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) \| 3 lines Remove extra debug message. ........ Merged revisions 397921-397922 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8300c9aaaf	Remove set but unused variable 'meid'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	00baddb906	Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	430bb3bfb3	Let Queue wrap up time influence member availability Queue members who happen to be in multiple queues at the same time may not have any wrap up time. This problem occurred due to a code change in Asterisk 11.3.0 that unified device state tracking of Queue members in multiple Queues (which fixed some other problems, but unfortunately caused this one). This patch fixes the behavior by having the is_member_available function check the queue's wrap up time and the time of the member's last call, such that for a particular queue, the member won't be considered available if their last call is within the wrap up time. (closes issue ASTERISK-22189) Reported by: Tony Lewis Tested by: Tony Lewis ........ Merged revisions 396948 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	59753b1ea1	Strip down the old event system This removes unused code, event types, IE pltypes, and event IE types where possible and makes several functions private that were once public. This includes a renumbering of the remaining event and IE types which breaks binary compatibility with previous versions. The last remaining consumers of the old event system (or parts thereof) are main/security_events.c, res/res_security_log.c, tests/test_cel.c, tests/test_event.c, main/cel.c, and the CEL backends. Review: https://reviewboard.asterisk.org/r/2703/ (closes issue ASTERISK-22139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e47d3db365	Doxygen comment tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	200ed6a405	Perform Ring-No-Answer checks before processing Hangup logic The rna() routine will raise a Stasis message involving both the caller and the agent. This doesn't work so well if we already hung up the agent channel, as the channel doesn't quite exist. Not surprisingly, this will crash. This patch properly runs the rna subroutine (performing all of the Ring-No-Answer logic) prior to hanging up the agent channel. (closes issue ASTERISK-22258) Reported by: Kiril Valchev Tested by: Kiril Valchev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c8a91b5b01	Add queue member paused hints This patch adds the ability in Queue to raise a hint when a member's paused state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name} are the name of the queue and the name of the member to subscribe to, respectively. For example: exten => 8501,hint,Queue:sales_pause_mark. Members will show as In Use when paused. Note that the format of the queue pause hint was changed slightly from what is on the issue to accomodate suggestion on the code review. Review: https://reviewboard.asterisk.org/r/2254 (closes issue ASTERISK-20842) Reported by: Philippe Lindheimer patches: qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch This patch renames the bridging* files to bridge. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9d8a5ceb02	Move after bridge callbacks into their own file One more major refactoring to go. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	684c83b29b	Add transfer support to CEL This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f54b1f2788	Provide error message for QUEUE_MEMBER when member is not in queue When QUEUE_MEMBER is used and the member specified is not in the queue, Asterisk provides an ERROR message that indicates that the option specified is not valid. This patch now properly displays an ERROR message that the member is not in the queue if an interface is specified. (closes issue ASTERISK-21980) Reported by: Avraam David ........ Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	909ee4bfb9	Refactor extraneous channel events This change removes JitterBufStats, ChannelReload, and ChannelUpdate and refactors the following events to travel over Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone * SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/ (closes issue ASTERISK-21476) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a1e219ef51	CEL refactoring cleanup This change removes AST_CEL_BRIDGE_UPDATE since it should no longer be used because masquerade situations are now accounted for in other ways. This also refactors usage of AST_CEL_FORWARD to be produced by a Dial message which has been extended with a "forward" field. (closes issue ASTERISK-21566) Review: https://reviewboard.asterisk.org/r/2635/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now very reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c2e29abcbf	Add announce-to-first-user option for app_queue In r386792, the ability to play prompts to the first caller in a call queue was added. While this is arguably a bug fix for those who expect the first caller to continue receiving prompts while the agent is dialed, it has the side effect of preventing the first caller from hearing the agent immediately upon bridging. This may not be a problem for those who really want this option, but for those who didn't care whether or not the first caller in queue heard their position, it was an issue. This patch disables the ability for the first caller in the queue to hear prompts and adds a new option, announce-to-first-user, to queues.conf. Those who the behavior can enable it by setting this value to True. Note that if we ever implement the ability to have the prompts be stopped upon bridging, this option can be removed. (closes issue ASTERISK-21782) Reported by: Remi Quezada ........ Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391241 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	a2d02edca5	Make app_queue AMI events more consistent. Give Join/Leave more useful names. This also removes the eventwhencalled and eventmemberstatus configuration options. These events can just be filtered via manager.conf blacklists. (closes issue ASTERISK-21469) Review: https://reviewboard.asterisk.org/r/2586/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	b6aac885be	Add dial events to app_queue and app_followme. Also fixes an issue in app_dial, where the channels were swapped on dial events. (closes issue ASTERISK-21551) (closes issue ASTERISK-21550) Review: https://reviewboard.asterisk.org/r/2549/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	908ac3507a	Conditional out more app_queue logging that needs to be reworked. Fixes crash because app_queue was unconditionally freeing a datastore that was still on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b97c71bb11	Fix shutdown assertions in stasis-core In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	bb52414990	Fix Segfault In app_queue When "persistentmembers" Is Enabled And Using Realtime When the "ignorebusy" setting was deprecated, we added some code to allow us to be compatible with older setups that are still using the "ignorebusy" setting instead of "ringinuse". We set a char *variable with the column name to use, which helps the realtime functions to use the correct column in their SQL queries. When "persistentmembers" is enabled, we are not setting this variable before the realtime functions were called to load members. This results in the variable being NULL and therefore causing a segfault when loading members during the module's process of loading. The solution was to move the code that sets that variable to be before these realtime functions are called during the loading of the module. (closes issue ASTERISK-21738) Reported by: JoshE Tested by: JoshE Patches: asterisk-21738-rt-ringinuse-field-not-set.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2499/ ........ Merged revisions 388108 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Olle Johansson	465d0f4a22	Play periodic prompts for first call in a call queue Review: https://reviewboard.asterisk.org/r/2263/ ........ Merged revisions 386792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386794 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	191cf99ae1	Move device state distribution to Stasis-core In the move from Asterisk's event system to Stasis, this makes distributed device state aggregation always-on, removes unnecessary task processors where possible, and collapses aggregate and non-aggregate states into a single cache for ease of retrieval. This also removes an intermediary step in device state aggregation. Review: https://reviewboard.asterisk.org/r/2389/ (closes issue ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	a7c5183d67	Fix Manager Segfault When app_queue Is Unloaded When app_queue is unloaded, some manager commands are not being unregistered which result in a segfault. This patch corrects this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann, Corey Farrell Tested by: Corey Farrell Patches: asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L. Young (license 5026) asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2444/ ........ Merged revisions 385593 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385594 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2e1e0735fe	Revamp of terminal color codes The core module related to coloring terminal output was old and needed some love. The main thing here was an attempt to get rid of the obscene number of stack-local buffers that were allocated for no other reason than to colorize some output. Instead, this uses a simple trick to allocate several buffers within threadlocal storage, then automatically rotates between them, so that you can make multiple calls to the colorization routine within one function and not need to allocate multiple buffers. Review: https://reviewboard.asterisk.org/r/2241/ Patches: bug.patch uploaded by Tilghman Lesher git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c41e70d647	app_queue: Fix incorrect assertion. (issue ASTERISK-16115) ........ Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8ed2c74fe3	app_queue: Fix multiple calls to a queue member that is in only one queue. When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	8c0d7005f3	Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ ........ Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8fb5bdce9a	Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0f54b3ee37	app_queue: Revert bad ringinuse=no patch. With the option ringinuse=no set, the patch committed for ASTERISK-16115 causes non-SIP queue members to never be called because the device state is checked after a channel is created to determine if the member is busy. These queue members always get the "Member %s is busy, cannot dial" message. Most channel drivers other than chan_sip use the default device state handling. The default device-state state is considered in use or unknown if the channel exists or not respectively. (closes issue ASTERISK-20801) Reported by: rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378038 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1c2f27c4a9	app_queue: Make update_status() not return anything. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f2bb9afe17	Multiple revisions 375993-375994 ........ r375993 \| mmichelson \| 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) \| 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 \| mmichelson \| 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) \| 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1eb14dbff8	Ensure that CDRs for a caller in a Queue that is not answered is NO ANSWER. When a caller enters a queue and no queue member answers the call, the current behaviour can be a little odd depending on the paused status of the queue members. If any queue member is paused, but not all, the CDR disposition will be BUSY. If all queue members are paused, then the CDR disposition is based instead on the disposition of the call prior to entering the Queue. This patch modifies the behaviour in the following ways: * If no queue members are paused, the CDR disposition is whatever the disposition was prior to going into Queue. If the call was answered this will be ANSWERED; otherwise, it is NO ANSWER. * If some queue members are pused, the CDR result is NO ANSWER. (This is a change in behaviour, as the result would previously have been BUSY) * If all queue members are paused, the CDR result is whatever the result was prior to going into Queue. This is the same as the behaviour prior to this patch. * If the caller hangs up, times out, or presses '*' with the 'h' option, the CDR disposition is again not set and is dependent on whether or not the caller was Answered prior to entering Queue. This patch was based on one provided by Thomas Arimont, but has been modified to accomodate findings by the reviewers. Review: https://reviewboard.asterisk.org/r/2064/ (closes issue AST-906) Reported by: Thomas Arimont (closes issue ASTERISK-17776) Reported by: Attila Megyeri git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	42a83618dd	app_queue: Make ordering of rrmemory/rrordered persist over add/remove members Prior to this patch, adding, removing or reloading members to rrmemory would cause the order to become completely jumbled. Now it behaves more or less like rrordered other than the fact that it stores the members on a hash table rather than a linked list. This patch also prevents removal of members and member reloads from jumbling rrordered queues. (issue AST-989) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged revisions 375216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375217 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375219 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	cfc6f60ca3	Doxygen Updates - Title update Update and extend the configuration_file group and enable linking to the application. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	516c9ec665	app_queue: Made pass connected line updates from the caller to ringing queue members. Party A calls Party B Party B puts Party A on hold. Party B calls a queue. Ringing queue member D sees Party B identification. Party B transfers Party A to the queue. Queue member D does not get a connected line update for Party A. Queue member D answers the call and still sees Party B information. However, if Party A later transfers the call to Party C then queue member D gets a connected line update for Party C. * Made pass connected line updates from the caller to queue members while the queue members are ringing. (closes issue AST-1017) Reported by: Thomas Arimont (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: rmudgett ........ Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 374802 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374803 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374804 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	14be2a5514	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b9eeff1521	app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374150 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	f8a37188f0	app_queue: 'agent available' hint, cleanup restart, and initial state Fix previously untested senarios; 1). On queue initialisation set queue_avail devstate to INUSE. Previously was unavailable, which indicated an agent was available. 2). When removing members, if there are no other members available, set queue_avail to INUSE. Previously, if a member interface had become 'unavailable', they were never going to be removed, particularly when persistant queues is enabled. 3). When adding a member, check that they are available, if they are set queue_avail to NOT_INUSE. Previously on reloaded, members may have been 'unavailable'. 4). When pausing or unpausing a member, set appropriate queue availability. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2129/ ........ Merged revisions 373804 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0a9d89d6be	"show" completion option for "queue" shouldn't appear twice When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts ........ Merged revisions 373666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373675 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373688 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	fd98835f1f	Doxygen Updates Janitor Work * Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	f56c0ecf9c	app_queue: Make queue reload members and variants of that work Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373300 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373318 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ca0e96ae19	Add queue monitoring hints This patch adds support for hints on a queue. Hints can be added using the nomenclature 'Queue:name', where name is the name of the queue being monitored. This nifty feature was done by Alec Davis. Review: https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis Tested by: alecdavis patches: review1619.diff2 by alecdavis (license 585) ........ Merged revisions 373235 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f1fb120f5d	Support all ways a member can be available for 'agent available' hints Alec's patch in r373188 added the ability to subscribe to a hint for when Queue members are available. This patch modifies the check that determines when a Queue member is available by refactoring the availability checks in num_available_members into a shared function is_member_available. This should now handle the ringinuse option, as well as device state values other than AST_DEVICE_NOT_INUSE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	67ca3b9126	app_queue: Support an 'agent available' hint Sets INUSE when no free agents, NOT_INUSE when an agent is free. modifes handle_statechange() scan members loop to scan for a free agent and updates the Queue:queuename_avial devstate. Previously exited early if the member was found in the queue. Now Exits later when both a member was found, and a free agent was found. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2121/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2a6be9fd0a	Fix exception path typo in app_queue.c try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper ........ Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372625 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372626 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5da59112b7	Update QueueMemberStatus event documentation to include member status values The Status: header in a QueueMemberStatus event (and other QueueMember* events) is the numeric value of the device state corresponding to that Queue Member. As those values are not exactly obvious, listing them in the documentation is useful. Matt Riddell reported this indirectly through the wiki page. (closes issue ASTERISK-20243) Reported by: Matt Riddell ........ Merged revisions 372531 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0090fb558d	Ensure listed queues are not offered for completion When using tab-completion for the list of queues on "queue reset stats" or "queue reload {all\|members\|parameters\|rules}", the tab-completion listing for further queues erroneously listed queues that had already been added to the list. The tab-completion listing now only displays queues that are not already in the list. (closes issue AST-963) Reported-by: John Bigelow ........ Merged revisions 372517 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372518 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372519 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c16141dda1	Ensure "rules" is tab-completable for "queue show" Previously, tabbing at the end of "queue show" produced a list of available queues about which information could be shown, but did not include an alternative command, "rules", to access information about queue rules. The "rules" item should now be shown in the list of tab-completable items. (closes issue AST-958) Reported-by: John Bigelow ........ Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372445 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372446 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	b02c65752c	app_queue: Only log PAUSEALL/UNPAUSEALL when 1+ memebers changed. Prior to this patch, if pause or unpause was issued on an interface without specifying a specific queue, a PAUSEALL or UNPAUSEALL event would be logged in the queue log even if that interface wasn't a member of any queues. This patch changes it so that these events are only logged when at least one member of any queue exists for that interface. (closes issue AST-946) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2079/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	1ab2639cf2	Prevent crash on shutdown due to refcount error on queues container. When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. ........ Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372090 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372091 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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