27171 Commits (0aa47e11e1faad77f6e2dabc25e34784ff0edb6f)

 

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Author	SHA1	Message	Date
Joshua Colp	eb3d083551	Merge "cel/cel_radius: Fix wrong pointer." into 13	10 years ago
Joshua Colp	b1ee692568	Merge topic 'pbx-split' into 13 ... * changes: main/pbx: Move hangup handler routines to pbx_hangup_handler.c. main/pbx: Move dialplan application management routines to pbx_app.c. main/pbx: Move switch routines to pbx_switch.c.	10 years ago
Joshua Colp	2e61c3d97b	Merge "main/pbx: Move timing routines to pbx_timing.c." into 13	10 years ago
Aaron An	86eae38d7e	cel/cel_radius: Fix wrong pointer. ... The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter y not the address of y. I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, then i review the source code and compare it with cdr/cdr_radius.c. Fix it and it works. ASTERISK-25647 #close Reported by: Aaron An Tested by: Aaron An Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0	10 years ago
George Joseph	881dc862e0	asterisk.h: Add ASTERISK_REGISTER_FILE macro ... The 11/13 branches and master use 2 different file version macros. 11/13 uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This means a new file added to 11/13 can't just be cherry-picked to master because the macro has to be changed. To make cherry-picking possible, ASTERISK_REGISTER_FILE was added to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL) The "$Revision$" tag doesn't do anything since Asterisk moved to git so just passing NULL as the verison works fine. asterisk.h was also annotated to deprecate ASTERISK_FILE_VERSION and suggest using ASTERISK_REGISTER_FILE for all new files. Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c, were modified to use the new macro to make sure it actually worked. 'core show file version' showed the correct output. Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5	10 years ago
Matt Jordan	9016e51c03	Merge "main/pbx: Move variable routines to pbx_variables.c." into 13	10 years ago
George Joseph	d228b62fd4	stasis_cache_pattern: Backport to 13 ... Somehow stasis_cache_pattern got out of sync between 13 and master and it was causing duplicate channel message issues in 13 when related to a specific endpoint. I.E. from statsd, 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice. Backporting stasis_cache_pattern from master to 13 solved the issue and running the unit and testsuite tests confirmed that no new ones were created. ASTERISK-25317 #close Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548	10 years ago
Corey Farrell	e462f0063f	main/pbx: Move hangup handler routines to pbx_hangup_handler.c. ... This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves hangup handler management functions to their own source. Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104	10 years ago
Corey Farrell	ab191d124c	main/pbx: Move dialplan application management routines to pbx_app.c. ... This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves dialplan application management functions to their own source. Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c	10 years ago
Corey Farrell	09a9b93896	main/pbx: Move switch routines to pbx_switch.c. ... This is the fifth patch in a series meant to reduce the bulk of pbx.c. This moves ast_switch functions to their own source. Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e	10 years ago
Corey Farrell	c608274a39	main/pbx: Move timing routines to pbx_timing.c. ... This is the fourth patch in a series meant to reduce the bulk of pbx.c. This moves pbx timing functions to their own source. Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6	10 years ago
Martin Tomec	338a8ffed6	app_queue: Add member flag "in_call" to prevent reading wrong lastcall time ... Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500	10 years ago
Rodrigo Ramírez Norambuena	e13719bff1	app_queue: Added reason pause of member ... In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e	10 years ago
Joshua Colp	937ba51659	Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime" into 13	10 years ago
George Joseph	4ec85a9f07	voicemail: Move app_voicemail / res_mwi_external conflict to runtime ... The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247	10 years ago
Corey Farrell	7fdcfd7724	main/pbx: Move variable routines to pbx_variables.c. ... This is the third patch in a series meant to reduce the bulk of pbx.c. This moves channel and global variable routines to their own source. Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6	10 years ago
Richard Mudgett	80a8b2a4cd	app_dial: Immediately exit dial if the caller is already hung up. ... If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418	10 years ago
Matt Jordan	f42036bf6b	Merge "main/cdr: Allow setting properties on a finalized CDR if it is the last one" into 13	10 years ago
Matt Jordan	4e1d9c4a20	Merge "main/cdr: Set the end time on a CDR if endbeforehexten is Yes" into 13	10 years ago
Matt Jordan	1087b0c6ed	main/cdr: Allow setting properties on a finalized CDR if it is the last one ... Prior to this patch, we explicitly disallowed setting any properties on a finalized CDR. This seemed like a good idea at the time; in practice, it was more restrictive. There are weird and strange scenarios where setting a property on a finalized CDR is definitely wrong. For example, we may Fork a CDR, finalizing the previous one, then change a property. In said case, the old CDR is supposed to now be 'immutable' (so to speak), and should not be updated. From the perspective of the code, a forked CDR that is finalized is just finalized. Hence why we decided these should not be updated. In practice, it is much more common to want to set a property on a CDR in the h extension or in a hangup handler. Disallowing a common scenario to make an esoteric behaviour work isn't good. This patch fixes this by allowing callers to set a property IF we are the last CDR in the chain. This preserves the finalized CDR if it was forked, while allowing the more common case to function. ASTERISK-25458 #close Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9	10 years ago
Matt Jordan	1f23e65b89	main/cdr: Set the end time on a CDR if endbeforehexten is Yes ... Prior to this patch, the CDR engine attempted to set the end time on a CDR that was executing hangup logic and with endbeforehexten set to Yes by calling a function that inspects the properties on the Party A snapshot to determine if we are ready to set the end time. That always failed. This is because a Party A snapshot is not updated for CDRs that are executing hangup logic with endbeforehexten=Yes. Instead of calling a function that looks at the Party A snapshot, we just simply set the end time on the CDR. This is safe to call multiple times, and is safe to call at this point as we know that (a) we are executing hangup logic, and (b) we are supposed to set the end time at this point. ASTERISK-25458 Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3	10 years ago
Corey Farrell	2ffade4574	main/pbx: Move custom function routines to pbx_functions.c. ... This is the second patch in a series meant to reduce the bulk of pbx.c. This moves custom function management routines to their own source. Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177	10 years ago
Matt Jordan	bc7c882326	Merge "main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c" into 13	10 years ago
Matt Jordan	5a75caa9e6	Merge "res_pjsip_history: Add a module that provides PJSIP history for debugging" into 13	10 years ago
George Joseph	20b8474f20	main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c ... We joked about splitting pbx.c into multiple files but this first step was fairly easy. All of the pbx_builtin dialplan applications have been moved into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() is called by asterisk.c just after load_pbx(). A few functions were renamed and are cross-exposed between the 2 source files. Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a	10 years ago
Joshua Colp	a68467d293	Merge "res_http_websocket.c: prevent avoidable disconnections caused by write errors" into 13	10 years ago
Joshua Colp	47d508e911	Merge "test_time: Provide a timeout when waiting." into 13	10 years ago
Matt Jordan	e4a566918a	tests/test_stasis_endpoints: Remove expected duplicate events ... The cache_clear test was written to expect duplicate Stasis messages sent from the technology endpoint to the all caching topic. This patch fixes the test to no longer expect these duplicate messages. ASTERISK-25137 Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981	10 years ago
Joshua Colp	a280400758	test_time: Provide a timeout when waiting. ... The test_timezone_watch unit test is written to expect a condition to be signaled when the inotify daemon thread runs. There exists a small window where the test_timezone_watch thread can signal the inotify daemon thread while it is not reading on the underlying file descriptor. If this occurs the test_timezone_watch thread will wait indefinitely for a signal that will never arrive. This change adds a timeout to the condition so it will return regardless after a period of time. Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390	10 years ago
George Joseph	3a1c4885be	endpoint/stasis: Eliminate duplicate events on endpoint status change ... When an endpoint is created, its messages are forwarded to both the tech endpoint topic and the all endpoints topic. This is done so that various parties interested in endpoint messages can subscribe to just the tech endpoint and receive all messages associated with that particular technology, as opposed to subscribing to the all endpoints topic. Unfortunately, when the tech endpoint is created, it also forwards all of its messages to the all topic. This results in duplicate messages whenever an endpoint publishes its messages. This patch resolves the duplicate message issue by creating a new function for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts as a normal caching topic, save that it no longer forwards messages it receives to the all endpoints topic. This allows it to act as an aggregation "sink", while preserving the necessary caching behaviour. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b	10 years ago
Dade Brandon	136c537695	res_http_websocket.c: prevent avoidable disconnections caused by write errors ... Updated ast_websocket_write to encode the entire frame in to one write operation, to ensure that we don't end up with a situation where the websocket header has been sent, while the body can not be written. Previous to August's patch in commit b9bd3c14, certain network conditions could cause the header to be written, and then the sub-sequent body to fail - which would cause the next successful write to contain a new header, and a new body (resulting in the peer receiving two headers - the second of which would be read as part of the body for the first header). This was patched to have both write operations individually fail by closing the websocket. In a case available to the submitter of this patch, the same body which would consistently fail to write, would succeed if written at the same time as the header. This update merges the two operations in to one, adds debug messages indicating the reason for a websocket connection being closed during a write operation, and clarifies some variable names for code legibility. Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598	10 years ago
Matt Jordan	5e99ba393e	Merge "bridge_basic.c: Fix GOTO_ON_BLINDXFR" into 13	10 years ago
Matt Jordan	1a1f64f36f	Merge "chan_sip.c: fix websocket_write_timeout default value" into 13	10 years ago
Joshua Colp	efa0868295	Merge "res/res_pjsip_location: Delete contact_status object when contact is deleted" into 13	10 years ago
Matt Jordan	84b1475674	Merge "Remove res_jabber file that was left behind." into 13	10 years ago
Corey Farrell	f2efbb5d75	Remove res_jabber file that was left behind. ... Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b	10 years ago
Matt Jordan	dde7f3c1c4	res_pjsip_history: Add a module that provides PJSIP history for debugging ... This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36	10 years ago
Dade Brandon	be050f2638	chan_sip.c: fix websocket_write_timeout default value ... websocket_write_timeout was not being set to its default value during sip config reload, which meant that prior to this commit, 1) the default value of 100 was not used, unless an invalid value (or 1) was specified in sip.conf for websocket_write_timeout, and 2) if the websocket_write_timeout directive was removed from sip.conf without a full restart of asterisk, then the previous value would continue to be used indefinitely. This essentially lead to a 0ms write timeout (the first write attempt in ast_careful_fwrite must have succeeded) in websocket write requests from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. Changes to websocket_write_timeout still only apply to new websocket sessions, after the sip reload -- timeouts on existing sessions are not adjusted during sip reload. Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953	10 years ago
Richard Mudgett	b3024cad10	bridge_basic.c: Fix GOTO_ON_BLINDXFR ... Use of GOTO_ON_BLINDXFR would not work at all. The target location would never be executed by the transferring channel. * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with valid context, exten, and priority parameters from the transferring channel. * Renamed some feature_blind_transfer() local variables for clarity. ASTERISK-25641 #close Reported by Dmitry Melekhov Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac	10 years ago
Matt Jordan	0a9941de9d	res/res_pjsip_location: Delete contact_status object when contact is deleted ... In 450579e908, a change was made that removed the deletion of the 'contact_status' object when a 'contact' object is deleted in sorcery. This unfortunately means that the 'contact_status' object persists, even when something has explicitly removed a contact. The result is that the state of the contact will not be regenerated if that contact is re-created, and the stale state will be reported/used for that contact. It also results in no ContactStatusChanged events being generated for either ARI or AMI. This patch restores the deletion logic that was removed. Doing so now results in the expected events being generated again. Change-Id: I28789a112e845072308b5b34522690e3faf58f07	10 years ago
Kevin Harwell	1e24a0ca8a	res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef ... Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151	10 years ago
Matt Jordan	15641cb6cd	Merge "app_amd: Correct documentation to reflect functionality" into 13	10 years ago
Dade Brandon	1d3d20dd68	app_amd: Correct documentation to reflect functionality ... Update documentation to reflect that maximum_number_of_words has functionality inconsistent with the variable name (and inconsistent with prior documentation.) Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093	10 years ago
Dade Brandon	965a0eee46	res_rtp_asterisk: Resolve further timing issues with DTLS negotiation ... Resolves an edge case dtls negotiation delay for certain networks which somehow manage to drop the rtcp side's packet when these are both sent ast_rtp_remote_address_set, causing it to have to time-out and restart the handshake. Move dtls pending bio flush in to it's own function, and call it from ast_rtp_on_ice_complete, when we're rtp->ice, rather than when ast_rtp_remote_address_set. Keep the existing flush from the recent change to res_rtp_remote_address_set if ice is not being used. ASTERISK-25614 #close Reported-by: XenCALL Tested by: XenCALL Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5	10 years ago
Matt Jordan	7ec9b5b98f	Merge "app_queue: update RT members when the 1st call joins a queue with no agents" into 13	10 years ago
Joshua Colp	e652a786d6	Merge "json: Audit ast_json_* usage for thread safety." into 13	10 years ago
Mark Michelson	03dab00d1d	Merge topic 'alembic_fixes' into 13 ... * changes: Alembic: Increase column size of PJSIP AOR "contact". Alembic: Add PJSIP global keep_alive_interval.	10 years ago
Carlos Oliva	ae428d8460	app_queue: update RT members when the 1st call joins a queue with no agents ... If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682	10 years ago
Joshua Colp	62e49a7e20	Merge "res_sorcery_memory_cache: Add support for a full backend cache." into 13	10 years ago
Joshua Colp	59d5bb0613	res_sorcery_memory_cache: Add support for a full backend cache. ... This change introduces the configuration option 'full_backend_cache' which changes the cache to be a full mirror of the backend instead of a per-object cache. This allows all sorcery retrieval operations to be carried out against it and is useful for object types which are used in a "retrieve all" or "retrieve some" pattern. ASTERISK-25625 #close Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5	10 years ago