asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	656bd42b11	app: Queue hangup if channel is hung up during sub or macro execution. This issue was exposed when executing a connected line subroutine. When connected or redirected subroutines or macros are executed it is expected that the underlying applications and logic invoked are fast and do not consume frames. In practice this constraint is not enforced and if not adhered to will cause channels to continue when they shouldn't. This is because each caller of the connected or redirected logic does not check whether the channel has been hung up on return. As a result the the hung up channel continues. This change makes it so when the API to execute a subroutine or macro is invoked the channel is checked to determine if it has hung up. If it has then a hangup is queued again so the caller will see it and stop. ASTERISK-25690 #close Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea	10 years ago
Sean Bright	2d29e67567	res_musiconhold: Prevent multiple simultaneous reloads. There are two ways in which the reload() function in res_musiconhold can be called from the CLI: * module reload res_musiconhold.so * moh reload In the former case, the module loader holds a lock that prevents multiple concurrent calls, but in the latter there is no such protection. This patch changes the 'moh reload' CLI command to invoke the module loader directly, rather than call reload() explicitly. ASTERISK-25687 #close Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c	10 years ago
Joshua Colp	71986dc85b	Merge topic 'update_taskprocessor_commands' into 11 * changes: taskprocessor.c: Fix CLI "core show taskprocessors" unref. taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock.	10 years ago
Joshua Colp	22238ee35b	Merge "ccss.c: Replace space in taskprocessor name." into 11	10 years ago
Joshua Colp	2bdb04131f	Merge "app_queue: Add member flag "in_call" to prevent reading wrong lastcall time" into 11	10 years ago
Joshua Colp	33407a66c4	Merge "pbx: Deadlock between contexts container and context_merge locks" into 11	10 years ago
Joshua Colp	524cf33e51	Merge "pbx_dundi: Run cleanup on failed load." into 11	10 years ago
Joshua Colp	2012acac20	Merge "res_crypto: Perform cleanup at shutdown." into 11	10 years ago
Joshua Colp	baee1edc44	Merge "res_calendar: Cleanup scheduler context at unload." into 11	10 years ago
Joshua Colp	186922d8c1	Merge "manager: Cleanup manager_channelvars during shutdown." into 11	10 years ago
Kevin Harwell	6744b09a99	pbx: Deadlock between contexts container and context_merge locks Recent changes (ASTERISK-25394 commit `2bd27d1222`) introduced the possibility of a deadlock. Due to the mentioned modifications ast_change_hints now needs to keep both merge/delete and state callbacks from occurring while it executes. Unfortunately, sometimes ast_change_hints can be called with the contexts container locked. When this happens it's possible for another thread to grab the context_merge_lock before the thread calling into ast_change_hints does and then try to obtain the contexts container lock. This of course causes a deadlock between the two threads. The thread calling into ast_change_hints waits for the other thread to release context_merge_lock and the other thread is waiting on that one to release the contexts container lock. Unfortunately, there is not a great way to fix this problem. When hints change, the subsequent state callbacks cannot run at the same time as a merge/delete, nor when the usual state callbacks do. This patch alleviates the problem by having those particular callbacks (the ones run after a hint change) occur in a serialized task. By moving the context_merge_lock to a task it can now safely be attempted or held without a deadlock occurring. ASTERISK-25640 #close Reported by: Krzysztof Trempala Change-Id: If2210ea241afd1585dc2594c16faff84579bf302	10 years ago
Corey Farrell	b41d4ce58e	devicestate: Cleanup engine thread during graceful shutdown. ASTERISK-25681 #close Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1	10 years ago
Corey Farrell	ef67dc11bd	manager: Cleanup manager_channelvars during shutdown. ASTERISK-25680 #close Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446	10 years ago
Corey Farrell	006ce3679c	res_calendar: Cleanup scheduler context at unload. ASTERISK-25679 #close Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f	10 years ago
Corey Farrell	656f7430f8	pbx_dundi: Run cleanup on failed load. During failed startup of pbx_dundi no cleanup was performed. Add a call to unload_module before returning AST_MODULE_LOAD_DECLINE. ASTERISK-25677 #close Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29	10 years ago
Corey Farrell	d2fd720290	res_crypto: Perform cleanup at shutdown. This change causes res_crypto to unregister CLI at shutdown while still preventing the module from being unloaded. ASTERISK-25673 #close Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc	10 years ago
Diederik de Groot	7188c72a07	include/asterisk/time.h: Renamed global declaration:tv Renamed global declaration:tv to dummy_tv_var_for_types, which would oltherwise cause 'shadow' warnings when 'tv' was declared as a local variable elsewhere. Added comment to note that dummy_tv_var_for_types is never really exported and only used as a place holder. ASTERISK-25627 #close Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28	10 years ago
Richard Mudgett	48dd99d2b4	taskprocessor.c: Fix CLI "core show taskprocessors" unref. Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5	10 years ago
Richard Mudgett	b6c4746409	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	10 years ago
Richard Mudgett	32cbf1e429	taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b	10 years ago
Joshua Colp	ea43b40c03	Merge "cel/cel_radius: Fix wrong pointer." into 11	10 years ago
Aaron An	2b0747817f	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	594b6a426b	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. Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5	10 years ago
Martin Tomec	1943cfc53c	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
Richard Mudgett	6bbb59e348	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
Joshua Colp	cc25230e0f	Merge "res_http_websocket.c: prevent avoidable disconnections caused by write errors" into 11	10 years ago
Joshua Colp	c8499b8d5a	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
Dade Brandon	10435bfa69	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
Dade Brandon	6929d2a428	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
Kevin Harwell	3066b80259	res_rtp_asterisk: rtp->ice check not wrapped in USE_PJPROJECT ifdef Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151	10 years ago
Matt Jordan	5e7fc1187f	Merge "app_amd: Correct documentation to reflect functionality" into 11	10 years ago
Dade Brandon	4dd42c1cfa	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	6dc4aacaed	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
Carlos Oliva	54f6f54a5f	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
Matt Jordan	e0beaf4e9b	Merge "AMI: Fixed OriginateResponse message" into 11	10 years ago
pchero	b9f9ffb351	AMI: Fixed OriginateResponse message When asterisk sends the OriginateResponse event message, it doesn't send the correct information for origination to an application. ASTERISK-25624 #close Change-Id: I8fb6bc533d7f812c66a24cc2b0354ddb09d0f640	10 years ago
server-pandora	5e6b1476a0	res_rtp_asterisk.c: Fix DTLS negotiation delays. - Trigger pending DTLS packets to send out, once the RTP instance's remote address is set. - Avoids locking the DTLS structure unnecessarily by only doing this if DTLS is passive. - Add DTLS locks around the structurally sensitive calls in the SSL portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock inside of itself, and we're dealing with the SSL BIO in at least two threads. WebRTC channels may receive a DTLS handshake before ast_rtp_remote_address_set is called, which causes there to be a pending response to send out. Previous to 1ad827, this was handled by calling dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP packet could trigger the pending handshake response. Since that was rightfully removed, whenever the DTLS handshake is received before the remote address is set, we would have to wait until another SSL packet arrives. As of Chrome M47's optimizations to their handshake process, WebRTC conversations between Chrome M47+ and Asterisk, where Asterisk is passive, experience a 1 second delay without this patch, because the SSL handshake is received before ICE negotation stores the remote_address, and the next SSL packet isn't received until after a 1 second timeout in Chrome, which causes a new handshake request. ASTERISK-25614 #close Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908	10 years ago
Matt Jordan	0198aeefab	Merge "main/utils: Don't emit an ERROR message if the read end of a pipe closes" into 11	10 years ago
Matt Jordan	d4e3491293	main/utils: Don't emit an ERROR message if the read end of a pipe closes An ERROR or WARNING message should generally indicate that something has gone wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not in control of when the far end closes its reading on a file descriptor. If the far end does close the file descriptor in an unclean fashion, this isn't a bug or error in Asterisk, particularly when the situation can be gracefully handled in Asterisk. Currently, when this happens, a user would see the following somewhat cryptic ERROR message: "utils.c: write() returned error: Broken pipe" There's a few problems with this: (1) It doesn't provide any context, other than 'something broke a pipe' (2) As noted, it isn't actually an error in Asterisk (3) It can get rather spammy if the thing breaking the pipe occurs often, such as a FastAGI server (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even mask legitimate issues This patch changes ast_carefulwrite to only log an ERROR if we actually had one that was reasonably under our control. For debugging purposes, we still emit a debug message if we detect that the far side has stopped reading. Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566	10 years ago
Jonathan Rose	e1d08a39c5	chan_sip: Add TCP/TLS keepalive to TCP/TLS server Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36	10 years ago
Corey Farrell	fa7329b72d	app_meetme: Set default value for audio_buffers. The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44	10 years ago
Filip Jenicek	62574fb9dc	chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) Asterisk may crash when calling ast_channel_get_t38_state(c) on a locked channel which is being hung up. ASTERISK-25609 #close Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b	10 years ago
Joshua Colp	640160a351	Merge "chan_sip.c: Start ICE negotiation when response is sent or received." into 11	10 years ago
Eugene Voityuk	badac7c340	chan_sip.c: Start ICE negotiation when response is sent or received. The current logic for ICE negotiation starts it when receiving an SDP with ICE candidates. This is incorrect as ICE negotiation can only start when each call party have at least one pair of local and remote candidate. Starting ICE negotiation early would result in negotiation failure and ultimately no audio. This change makes it so ICE negotiation is only started when a response with SDP is received or when a response with SDP is sent. ASTERISK-24146 Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca	10 years ago
Joshua Colp	0e303df9ad	Merge "translate: Avoid a warning message when doing FEC within Opus Codec." into 11	10 years ago
Alexander Traud	a388ff29e7	codec_resample: Increase buffer for Opus Codec with FEC. ASTERISK-25599 #close Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e	10 years ago
Alexander Traud	ac8e285dfd	translate: Avoid a warning message when doing FEC within Opus Codec. ASTERISK-25616 #close Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319	10 years ago
Richard Mudgett	47118fb947	chan_sip: Fix crash involving the bogus peer during sip reload. A crash happens sometimes when performing a CLI "sip reload". The bogus peer gets refreshed while it is in use by a new call which can cause the crash. * Protected the global bogus peer object with an ao2 global object container. ASTERISK-25610 #close Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed	10 years ago
Matt Jordan	9ca3d54203	Merge "build: Fix building with newer GCC." into 11	10 years ago
Joshua Colp	b59b0bb509	build: Fix building with newer GCC. Newer GCC is upset that the features flags are uninitialized. Zero them out so it is less upset. Change-Id: I05162699e1b36bce4145f58a63b47ad19c6975ac	10 years ago

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