27351 Commits (71bb7b9c401d4e1fc9bdfea65948de3c82bb08d2)

 

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Author	SHA1	Message	Date
Richard Mudgett	71bb7b9c40	taskprocessor.c: Sort CLI "core show taskprocessors" output. ... Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e	11 years ago
Richard Mudgett	84c245d38c	taskprocessor.c: Fix CLI "core show taskprocessors" output format. ... Update the CLI "core show taskprocessors" output format to not be distorted because UUID names are longer than previously used taskprocessor names. Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601	11 years ago
Richard Mudgett	7d86979ea0	taskprocessor.c: Fix CLI "core show taskprocessors" unref. ... Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5	11 years ago
Richard Mudgett	b025e1982f	taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. ... Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b	11 years ago
Richard Mudgett	1fb39aa8a0	ccss.c: Replace space in taskprocessor name. ... The CLI "core ping taskprocessor" command does not work very well with taskprocessor names that have spaces in them. You have to put quotes around the name so using tab completion becomes awkward. Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0	11 years ago
Mark Michelson	a146bde1f7	Merge "main: Use ast_strdup instead of strdup"	11 years ago
Mark Michelson	1afc8432dc	PJSIP: Prevent deadlock due to dialog/transaction lock inversion. ... A deadlock was observed where the monitor thread was stuck, therefore resulting in no incoming SIP traffic being processed. The problem occurred when two 200 OK responses arrived in response to a terminating NOTIFY request sent from Asterisk. The first 200 OK was dispatched to a threadpool worker, who locked the corresponding transaction. The second 200 OK arrived, resulting in the monitor thread locking the dialog. At this point, the two threads are at odds, because the monitor thread attempts to lock the transaction, and the threadpool thread loops attempting to try to lock the dialog. In this case, the fix is to not have the monitor thread attempt to hold both the dialog and transaction locks at the same time. Instead, we release the dialog lock before attempting to lock the transaction. There have also been some debug messages added to the process in an attempt to make it more clear what is going on in the process. ASTERISK-25668 #close Reported by Mark Michelson Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a	11 years ago
Corey Farrell	5d8c42c6d3	ast_format_cap_append_by_type: Resolve codec reference leak. ... This resolves a reference leak caused by ASTERISK-25535. The pointer returned by ast_format_get_codec is saved so it can be released. ASTERISK-25664 #close Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec	11 years ago
Diederik de Groot	7856762f2f	main: Use ast_strdup instead of strdup ... Fix compile error in main/utils.c because strdup was used in dummy_start Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793	11 years ago
Joshua Colp	0f87b89b32	Merge "cel/cel_radius: Fix wrong pointer."	11 years ago
Joshua Colp	9cded6eb36	Merge "Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts."	11 years ago
Walter Doekes	64b2046f3d	Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. ... The spandspflow2pcap.py creates pcap files from fax.log files, generated through 'fax set debug on' when receiving a fax. An example fax.log is included as spandspflow2pcap.log. The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a recent version of SIPp. ASTERISK-25660 #close Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526	11 years ago
Joshua Colp	c8e786ff66	Merge topic 'pbx-split' ... * 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.	11 years ago
Joshua Colp	e00e6a606e	Merge "main/pbx: Move timing routines to pbx_timing.c."	11 years ago
Aaron An	084563e136	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	11 years ago
Matt Jordan	f3a052667e	Merge "main/pbx: Move variable routines to pbx_variables.c."	11 years ago
Corey Farrell	36f1eaf0b5	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	11 years ago
Matt Jordan	193ac48db0	Merge "Happy new year 2016."	11 years ago
Joshua Colp	0c62e0a55c	Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime"	11 years ago
Corey Farrell	3507494b8a	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	11 years ago
Corey Farrell	54a8f1a396	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	11 years ago
Corey Farrell	c3c8b8e41d	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	11 years ago
George Joseph	6d18fe151c	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	11 years ago
Corey Farrell	5ee5c3739e	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	11 years ago
Richard Mudgett	f88b952093	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	11 years ago
Matt Jordan	839f328601	Merge "main/cdr: Allow setting properties on a finalized CDR if it is the last one"	11 years ago
Matt Jordan	5201984ed6	Merge "main/cdr: Set the end time on a CDR if endbeforehexten is Yes"	11 years ago
Matt Jordan	e9dd16364e	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	11 years ago
Matt Jordan	153547a9b1	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	11 years ago
Corey Farrell	f9bfc2450e	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	11 years ago
Matt Jordan	11d99e0809	Merge "main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c"	11 years ago
Rodrigo Ramírez Norambuena	3fd528dddf	Happy new year 2016. ... Change-Id: I22d3c90f6f27df82e915bbf81c1d91221f7a945e	11 years ago
Matt Jordan	d49cc32017	Merge "res_pjsip_history: Add a module that provides PJSIP history for debugging"	11 years ago
Matt Jordan	9cdf3ec19d	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	11 years ago
George Joseph	5e67e51c6a	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	11 years ago
Joshua Colp	4e5250d5c2	Merge "res_http_websocket.c: prevent avoidable disconnections caused by write errors"	11 years ago
Joshua Colp	410c5ee0fc	Merge "test_time: Provide a timeout when waiting."	11 years ago
Matt Jordan	37f9050f77	Merge "endpoint/stasis: Eliminate duplicate events on endpoint status change"	11 years ago
Joshua Colp	a05bb258b1	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	11 years ago
Matt Jordan	96b32e0321	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	11 years ago
Dade Brandon	3bddcc0219	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	11 years ago
George Joseph	22db16fa81	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	11 years ago
Matt Jordan	b4b4a95225	Merge "bridge_basic.c: Fix GOTO_ON_BLINDXFR"	11 years ago
Matt Jordan	923b8cb913	Merge "chan_sip.c: fix websocket_write_timeout default value"	11 years ago
Joshua Colp	e3f1dbd91c	Merge "res/res_pjsip_location: Delete contact_status object when contact is deleted"	11 years ago
Corey Farrell	6b08f01c60	Remove res_jabber file that was left behind. ... Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b	11 years ago
Dade Brandon	6dc21bbf00	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	11 years ago
Richard Mudgett	8eb5da0679	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	11 years ago
Matt Jordan	2df4ad647c	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	11 years ago
Kevin Harwell	b8876711f3	res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef ... Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151	11 years ago