asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	07a850c6dd	app_playback.c: Fix PLAYBACKSTATUS regression. In Asterisk 11, if a channel was redirected away during Playback(), the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 (specifically commit `7d9871b394`) that behavior was inadvertently changed and the same operation would result in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 behavior has been restored. Partial fix for ASTERISK~25661. Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff	2 years ago
George Joseph	6c75383fd5	res_rtp_asterisk: Don't use double math to generate timestamps Rounding issues with double math were causing rtp timestamp slips in outgoing packets. We're now back to integer math and are getting no more slips. ASTERISK-30391 Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4	2 years ago
Igor Goncharovsky	de745157ca	res_pjsip_rfc3326: Add SIP causes support for RFC3326 Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	2 years ago
Alexei Gradinari	462133e5e4	format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...) Each playback of WAV files results in logging "Skipping unknown block 'LIST'". To prevent unnecessary flooding of this DEBUG log this patch replaces ast_log(LOG_DEBUG, ...) by ast_debug(1, ...). Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0	2 years ago
George Joseph	345ff2d8ee	res_rtp_asterisk: Asterisk Media Experience Score (MES) ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	8067229418	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `62745013a4`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I1665a5442bfb6d7bfa06fdcea3374f4581395b4a	2 years ago
Naveen Albert	bde2689e1b	loader: Allow declined modules to be unloaded. Currently, if a module declines to load, dlopen is called to register the module but dlclose never gets called. Furthermore, loader.c currently doesn't allow dlclose to ever get called on the module, since it declined to load and the unload function bails early in this case. This can be problematic if a module is updated, since the new module cannot be loaded into memory since we haven't closed all references to it. To fix this, we now allow modules to be unloaded, even if they never "loaded" in Asterisk itself, so that dlclose is called and the module can be properly cleaned up, allowing the updated module to be loaded from scratch next time. ASTERISK-30345 #close Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5	2 years ago
Naveen Albert	0dc3d8c655	app_broadcast: Add Broadcast application Adds a new application, Broadcast, which can be used for one-to-many transmission and many-to-one reception of channel audio in Asterisk. This is similar to ChanSpy, except it is designed for multiple channel targets instead of a single one. This can make certain kinds of audio manipulation more efficient and streamlined. New kinds of audio injection impossible with ChanSpy are also made possible. ASTERISK-30180 #close Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726	2 years ago
Naveen Albert	3e4c012215	func_frame_trace: Print text for text frames. Since text frames contain a text body, make FRAME_TRACE more useful for text frames by actually printing the text. ASTERISK-30353 #close Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50	2 years ago
Boris P. Korzun	e85f23e6e5	http.c: Fix NULL pointer dereference bug If native HTTP is disabled but HTTPS is enabled and status page enabled too, Core/HTTP crashes while loading. 'global_http_server' references to NULL, but the status page tries to dereference it. The patch adds a check for HTTP is enabled. ASTERISK-30379 #close Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249	2 years ago
Naveen Albert	811584ded3	json.h: Add ast_json_object_real_get. json.h contains macros to get a string and an integer from a JSON object. However, the macro to do this for JSON reals is missing. This adds that. ASTERISK-30361 #close Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a	2 years ago
Holger Hans Peter Freyther	6dcb908025	res_http_media_cache: Do not crash when there is no extension Do not crash when a URL has no path component as in this case the ast_uri_path function will return NULL. Make the code cope with not having a path. The below would crash > media cache create http://google.com /tmp/foo.wav Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault. 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 (gdb) bt #0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 #1 0x0000ffff43d43a78 in file_extension_from_string (str=<optimized out>, buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:288 #2 0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568, buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378 #3 0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392 #4 0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555 #5 0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>) at res_http_media_cache.c:613 #6 0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>) at bucket.c:191 #7 0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718, details=details@entry=0xffffca9974a8) at sorcery.c:2027 #8 0x0000000000559698 in ast_sorcery_create (sorcery=<optimized out>, object=object@entry=0x3bf96568) at sorcery.c:2077 #9 0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727 #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com", file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335 #11 0x00000000004f88ec in media_cache_handle_create_item (e=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8) at media_cache.c:640 ASTERISK-30375 #close Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273	2 years ago
Naveen Albert	9ede683f4e	manager: Fix appending variables. The if statement here is always false after the for loop finishes, so variables are never appended. This removes that to properly append to the end of the variable list. ASTERISK-30351 #close Reported by: Sebastian Gutierrez Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199	2 years ago
George Joseph	db6e9c4e50	res_pjsip_transport_websocket: Add remote port to transport When Asterisk receives a new websocket conenction, it creates a new pjsip transport for it and copies connection data into it. The transport manager then uses the remote IP address and port on the transport to create a monitor for each connection. However, the remote port wasn't being copied, only the IP address which meant that the transport manager was creating only 1 monitoring entry for all websocket connections from the same IP address. Therefore, if one of those connections failed, it deleted the transport taking all the the connections from that same IP address with it. * We now copy the remote port into the created transport and the transport manager behaves correctly. ASTERISK-30369 Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1	2 years ago
Naveen Albert	a29f3f864d	pbx_app: Update outdated pbx_exec channel snapshots. pbx_exec makes a channel snapshot before executing applications. This doesn't cause an issue during normal dialplan execution where pbx_exec is called over and over again in succession. However, if pbx_exec is called "one off", e.g. using ast_pbx_exec_application, then a channel snapshot never ends up getting made after the executed application returns, and inaccurate snapshot information will linger for a while, causing "core show channels", etc. to show erroneous info. This is fixed by manually making a channel snapshot at the end of ast_pbx_exec_application, since we anticipate that pbx_exec might not get called again immediately. ASTERISK-30367 #close Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086	2 years ago
George Joseph	62745013a4	res_rtp_asterisk: Asterisk Media Experience Score (MES) This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	40ed776c40	res_pjsip_session: Use Caller ID for extension matching. Currently, there is no Caller ID available to us when checking for an extension match when handling INVITEs. As a result, extension patterns that depend on the Caller ID are not matched and calls may be incorrectly rejected. The Caller ID is not available because the supplement that adds Caller ID to the session does not execute until after this check. Supplement callbacks cannot yet be executed at this point since the session is not yet in the appropriate state. To fix this without impacting existing behavior, the Caller ID number is now retrieved before attempting to pattern match. This ensures pattern matching works correctly and there is no behavior change to the way supplements are called. ASTERISK-28767 #close Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7	2 years ago
Ben Ford	cec98b5f0c	res_pjsip_sdp_rtp.c: Use correct timeout when put on hold. When a call is put on hold and it has moh_passthrough and rtp_timeout set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is expected to be used, but rtp_timeout is used instead. This change adds a couple of checks for locally_held to determine if rtp_timeout_hold needs to be used instead of rtp_timeout. ASTERISK-30350 Change-Id: I7b106fc244332014216d12bba851cefe884cc25f	2 years ago
Naveen Albert	1974ebe109	app_voicemail_odbc: Fix string overflow warning. Fixes a negative offset warning by initializing the buffer to empty. Additionally, although it doesn't currently complain about it, the size of a buffer is increased to accomodate the maximum size contents it could have. ASTERISK-30240 #close Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877	2 years ago
Igor Goncharovsky	997a2d70b9	res_pjsip: Fix path usage in case dialing with '@' Fix aor lookup on sip path addition. Issue happens in case of dialing with @ and overriding user part of RURI. ASTERISK-30100 #close Reported-by: Yury Kirsanov Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13	2 years ago
Peter Fern	b1f1556922	streams: Ensure that stream is closed in ast_stream_and_wait on error When ast_stream_and_wait returns an error (for example, when attempting to stream to a channel after hangup) the stream is not closed, and callers typically do not check the return code. This results in leaking file descriptors, leading to resource exhaustion. This change ensures that the stream is closed in case of error. ASTERISK-30198 #close Reported-by: Julien Alie Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803	2 years ago
Naveen Albert	914c8e28c1	func_callerid: Warn about invalid redirecting reason. Currently, if a user attempts to set a Caller ID related function to an invalid value, a warning is emitted, except for when setting the redirecting reason. We now emit a warning if we were unable to successfully parse the user-provided reason. ASTERISK-30332 #close Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a	2 years ago
Asterisk Development Team	d141f3a821	Update CHANGES and UPGRADE.txt for 18.16.0	2 years ago
Alexandre Fournier	70c650751e	res_geoloc: fix NULL pointer dereference bug The `ast_geoloc_datastore_add_eprofile` function does not return 0 on success, it returns the size of the underlying datastore. This means that the datastore will be freed and its pointer set to NULL when no error occured at all. ASTERISK-30346 Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9	2 years ago
Joshua C. Colp	b65aa359c4	res_pjsip_aoc: Don't assume a body exists on responses. When adding AOC to an outgoing response the code assumed that a body would exist for comparing the Content-Type. This isn't always true. The code now checks to make sure the response has a body before checking the Content-Type. ASTERISK-21502 Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76	2 years ago
Naveen Albert	462461870d	app_if: Fix format truncation errors. Fixes format truncation warnings in gcc 12.2.1. ASTERISK-30349 #close Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c	2 years ago
Naveen Albert	91a12a5cd4	res_pjsip_header_funcs: Add custom parameter support. Adds support for custom URI and header parameters in the From header in PJSIP. Parameters can be both set and read using this function. ASTERISK-30150 #close Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428	2 years ago
Michael Kuron	63c36ed9c4	manager: AOC-S support for AOCMessage ASTERISK-21502 Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08	2 years ago
Michael Kuron	d73f22268f	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333	2 years ago
Naveen Albert	833f2af3da	res_pjsip_session.c: Map empty extensions in INVITEs to s. Some SIP devices use an empty extension for PLAR functionality. Rather than rejecting these empty extensions, we now use the s extension for such calls to mirror the existing PLAR functionality in Asterisk (e.g. chan_dahdi). ASTERISK-30265 #close Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590	2 years ago
Marcel Wagner	fb281c48e2	res_pjsip: Fix typo in from_domain documentation This fixes a small typo in the from_domain documentation on the endpoint documentation ASTERISK-30328 #close Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21	2 years ago
Naveen Albert	3babc68099	res_hep: Add support for named capture agents. Adds support for the capture agent name field of the Homer protocol to Asterisk by allowing users to specify a name that will be sent to the HEP server. ASTERISK-30322 #close Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b	2 years ago
Naveen Albert	5a78ce5a1f	app_voicemail: Fix missing email in msg_create_from_file. msg_create_from_file currently does not dispatch emails, which means that applications using this function, such as MixMonitor, will not trigger notifications to users (only AMI events are sent our currently). This is inconsistent with other ways users can receive voicemail. This is fixed by adding an option that attempts to send an email and falling back to just the notifications as done now if that fails. The existing behavior remains the default. ASTERISK-30283 #close Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7	2 years ago
Naveen Albert	a7cd9ce26f	app_if: Adds conditional branch applications Adds the If, ElseIf, Else, ExitIf, and EndIf applications for conditional execution of a block of dialplan, similar to the While, EndWhile, and ExitWhile applications. The appropriate branch is executed at most once if available and may be broken out of while inside. ASTERISK-29497 Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49	2 years ago
Marcel Wagner	32be12e045	res_pjsip: Update contact_user to point out default Updates the documentation for the 'contact_user' field to point out the default outbound contact if no contact_user is specified 's' ASTERISK-30316 #close Change-Id: I61f24fb9164e4d07e05908a2511805281874c876	2 years ago
Naveen Albert	3d4162af44	res_adsi: Fix major regression caused by media format rearchitecture. The commit that rearchitected media formats, `a2c912e997` (ASTERISK_23114) introduced a regression by improperly translating code in res_adsi.c. In particular, the pointer to the frame buffer was initialized at the top of adsi_careful_send, rather than dynamically updating it for each frame, as is required. This resulted in the first frame being repeatedly sent, rather than advancing through the frames. This corrupted the transmission of the CAS to the CPE, which meant that CPE would never respond with the DTMF acknowledgment, effectively completely breaking ADSI functionality. This issue is now fixed, and ADSI now works properly again. ASTERISK-29793 #close Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe	2 years ago
Joshua C. Colp	f5a7efb963	ari: Destroy body variables in channel create. When passing a JSON body to the 'create' channel route it would be converted into Asterisk variables, but never freed resulting in a memory leak. This change makes it so that the variables are freed in all cases. ASTERISK-30344 Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d	2 years ago
Naveen Albert	30b4dc9bbb	func_presencestate: Fix invalid memory access. When parsing information from AstDB while loading, it is possible that certain pointers are never set, which leads to invalid memory access and then, fatally, invalid free attempts on this memory. We now initialize to NULL to prevent this. ASTERISK-30311 #close Change-Id: I6120681d04fd2c12a9473f35ce95a1f8e74e3929	2 years ago
Naveen Albert	f5ead740f3	sig_analog: Fix no timeout duration. ASTERISK_28702 previously attempted to fix an issue with flash hook hold timing out after just under 17 minutes, when it should have never been timing out. It fixed this by changing 999999 to INT_MAX, but it did so in chan_dahdi, which is the wrong place since ss_thread is now in sig_analog and the one in chan_dahdi is mostly dead code. This fixes this by porting the fix to sig_analog. ASTERISK-30336 #close Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15	2 years ago
Naveen Albert	86a40686e4	xmldoc: Allow XML docs to be reloaded. The XML docs are currently only loaded on startup with no way to update them during runtime. This makes it impossible to load modules that use ACO/Sorcery (which require documentation) if they are added to the source tree and built while Asterisk is running (e.g. external modules). This adds a CLI command to reload the XML docs during runtime so that documentation can be updated without a full restart of Asterisk. ASTERISK-30289 #close Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07	2 years ago
Naveen Albert	0fb48fb751	rtp_engine.h: Update examples using ast_format_set. This file includes some doxygen comments referencing ast_format_set. This is an obsolete API that was removed years back, but documentation was not fully updated to reflect that. These examples are updated to the current way of doing things (using the format cache). ASTERISK-30327 #close Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7	2 years ago
Naveen Albert	3cdc280cad	app_mixmonitor: Add option to use real Caller ID for voicemail. MixMonitor currently uses the Connected Line as the Caller ID for voicemails. This is due to the implementation being written this way for use with Digium phones. However, in general this is not correct for generic usage in the dialplan, and people may need the real Caller ID instead. This adds an option to do that. ASTERISK-30286 #close Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c	2 years ago
Ben Ford	cbb7585e7d	pjproject: 2.13 security fixes Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13. ASTERISK-30338 Change-Id: I86fdc003d5d22cb66e7cc6dc3313a8194f27eb69	2 years ago
George Joseph	5741569c5e	pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	2 years ago
Mike Bradeen	76d10ea668	manager: prevent file access outside of config dir Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404	2 years ago
Mike Bradeen	16a7a98643	ooh323c: not checking for IE minimum length When decoding q.931 encoded calling/called number now checking for length being less than minimum required. ASTERISK-30103 Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7	2 years ago
Maximilian Fridrich	37b0727288	core & res_pjsip: Improve topology change handling. This PR contains two relatively separate changes in channel.c and res_pjsip_session.c which ensure that topology changes are not ignored in cases where they should be handled. For channel.c: The function ast_channel_request_stream_topology_change only triggers a stream topology request change indication, if the channel's topology does not equal the requested topology. However, a channel could be in a state where it is currently "negotiating" a new topology but hasn't updated it yet, so the topology request change would be lost. Channels need to be able to handle such situations internally and stream topology requests should therefore always be passed on. In the case of chan_pjsip for example, it queues a session refresh (re-INVITE) if it is currently in the middle of a transaction or has pending requests (among other reasons). Now, ast_channel_request_stream_topology_change always indicates a stream topology request change even if the requested topology equals the channel's topology. For res_pjsip_session.c: The function resolve_refresh_media_states does not process stream state changes if the delayed active state differs from the current active state. I.e. if the currently active stream state has changed between the time the sip session refresh request was queued and the time it is being processed, the session refresh is ignored. However, res_pjsip_session contains logic that ensures that session refreshes are queued and re-queued correctly if a session refresh is currently not possible. So this check is not necessary and led to some session refreshes being lost. Now, a session refresh is done even if the delayed active state differs from the current active state and it is checked whether the delayed pending state differs from the current active - because that means a refresh is necessary. Further, the unit test of resolve_refresh_media_states was adapted to reflect the new behavior. I.e. the changes to delayed pending are prioritized over the changes to current active because we want to preserve the original intention of the pending state. ASTERISK-30184 Change-Id: Icd0703295271089057717006730b555b9a1d4e5a	2 years ago
Naveen Albert	84eb579052	pbx_builtins: Allow Answer to return immediately. The Answer application currently waits for up to 500ms for media, even if users specify a different timeout. This adds an option to not wait for media on the channel by doing a raw answer instead. The default 500ms threshold is also documented. ASTERISK-30308 #close Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4	2 years ago
Naveen Albert	0bbcda3040	chan_dahdi: Allow FXO channels to start immediately. Currently, chan_dahdi will wait for at least one ring before an incoming call can enter the dialplan. This is generally necessary in order to receive the Caller ID spill and/or distinctive ringing detection. However, if neither of these is required, then there is nothing gained by waiting for one ring and this unnecessarily delays call setup. Users can now use immediate=yes to make FXO channels (FXS signaled) begin processing dialplan as soon as Asterisk receives the call. ASTERISK-30305 #close Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5	2 years ago
Naveen Albert	3c7dde3bf6	sla: Prevent deadlock and crash due to autoservicing. SLAStation currently autoservices the station channel before creating a thread to actually dial the trunk. This leads to duplicate servicing of the channel which causes assertions, deadlocks, crashes, and moreover not the correct behavior. Removing the autoservice prevents the crash, but if the station hangs up before the trunk answers, the call hangs since the hangup was never serviced on the channel. This is fixed by not autoservicing the channel, but instead servicing it in the thread dialing the trunk, since it is doing so synchronously to begin with. Instead of sleeping for 100ms in a loop, we simply use the channel for timing, and abort if it disappears. The same issue also occurs with SLATrunk when a call is answered, because ast_answer invokes ast_waitfor_nandfds. Thus, we use ast_raw_answer instead which does not cause any conflict and allows the call to be answered normally without thread blocking issues. ASTERISK-29998 #close Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8	2 years ago

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33502 Commits (18) All Branches Search

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