asterisk

Commit Graph

Author	SHA1	Message	Date
MikeNaso	720813dc97	res_pjsip.c: Set contact_user on incoming call local Contact header If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls. Resolves: #226	2 years ago
Sean Bright	1171beb7e4	extconfig: Allow explicit DB result set ordering to be disabled. Added a new boolean configuration flag - `order_multi_row_results_by_initial_column` - to both res_pgsql.conf and res_config_odbc.conf that allows the administrator to disable the explicit `ORDER BY` that was previously being added to all generated SQL statements that returned multiple rows. Fixes: #179	2 years ago
George Joseph	21b0522abd	rest-api: Run make ari-stubs An earlier cherry-pick that involved rest-api somehow didn't include a comment change in res/ari/resource_endpoints.h. This commit corrects that. No changes other than the comment.	2 years ago
Naveen Albert	2179082eaf	res_pjsip_header_funcs: Make prefix argument optional. The documentation for PJSIP_HEADERS claims that prefix is optional, but in the code it is actually not. However, there is no inherent reason for this, as users may want to retrieve all header names, not just those beginning with a certain prefix. This makes the prefix optional for this function, simply fetching all header names if not specified. As a result, the documentation is now correct. Resolves: #230 UserNote: The prefix argument to PJSIP_HEADERS is now optional. If not specified, all header names will be returned.	2 years ago
Maximilian Fridrich	51a7b18038	core/ari/pjsip: Add refer mechanism This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.	2 years ago
Holger Hans Peter Freyther	f335da6b74	res_prometheus: Do not generate broken metrics In `8d6fdf9c3a` invisible bridges were skipped but that lead to producing metrics with no name and no help. Keep track of the number of metrics configured and then only emit these. Add a basic testcase that verifies that there is no '(NULL)' in the output. ASTERISK-30474	2 years ago
Sean Bright	c52b4ce11c	res_pjsip: Enable TLS v1.3 if present. Fixes #221 UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying PJSIP library. The bundled version of PJSIP supports TLS v1.3.	2 years ago
Sean Bright	fe467d595c	res_geolocation: Ensure required 'location_info' is present. Fixes #189	2 years ago
zhengsh	d3c4f93ca6	res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. From the gdb information, it was found that when calling __ast_free, the size of the allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb, it is found to be 1. Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid, which is outside the protection of the rtp_instance lock. However, ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses rtp->themssrc_valid within the protection of the rtp_instance lock. This can lead to the possibility that the value of rtp->themssrc_valid used in the call to ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used within ast_rtcp_generate_report(). Resolves: asterisk#63	2 years ago
Mike Bradeen	e84fe59cb2	res_musiconhold: avoid moh state access on unlocked chan Move channel unlock to after moh state access to avoid potential unlocked access to state. Resolves: #133	2 years ago
George Joseph	b974a8f9eb	rest-api: Ran make ari stubs to fix resource_endpoints inconsistency	2 years ago
Sean Bright	153095dd91	res_pjsip_rfc3326: Prefer Q.850 cause code over SIP. Resolves: #116	2 years ago
Ben Ford	61f37faf6d	res_pjsip_session: Added new function calls to avoid ABI issues. Added two new functions (ast_sip_session_get_dialog and ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the pjsip_inv_state respectively from the pjsip_inv_session on the ast_sip_session struct. This is due to pjproject adding a new field to the pjsip_inv_session struct that caused crashes when trying to access fields that were no longer where they were expected to be if a module was compiled against a different version of pjproject. Resolves: #145	2 years ago
Sean Bright	642b84c9c9	pjsip_transport_events.c: Use %zu printf specifier for size_t. Partially resolves #143.	2 years ago
Sean Bright	7e1f0b8e1f	res_crypto.c: Gracefully handle potential key filename truncation. Partially resolves #143.	2 years ago
zhou_jiajian	9b8311f9a6	res_fax_spandsp.c: Clean up a spaces/tabs issue	2 years ago
Sean Bright	b4d2e32831	res_crypto.c: Avoid using the non-portable ALLPERMS macro. ALLPERMS is not POSIX and it's trivial enough to not jump through autoconf hoops to check for it. Fixes #149.	2 years ago
Jiajian Zhou	73c5967c2d	AMI: Add parking position parameter to Park action Add a parking space extension parameter (ParkingSpace) to the Park action. Park action will attempt to park the call to that extension. If the extension is already in use, then execution will continue at the next priority. UserNote: New ParkingSpace parameter has been added to AMI action Park.	2 years ago
Naveen Albert	ce7a72d7e2	res_musiconhold: Add option to loop last file. Adds the loop_last option to res_musiconhold, which allows the last audio file in the directory to be looped perpetually once reached, rather than circling back to the beginning again. Resolves: #122 ASTERISK-30462 UserNote: The loop_last option in musiconhold.conf now allows the last file in the directory to be looped once reached.	2 years ago
Joe Searle	8462154a03	res_stasis.c: Add new type 'sdp_label' for bridge creation. Add new type 'sdp_label' when creating a bridge using the ARI. This will add labels to the SDP for each stream, the label is set to the corresponding channel id. Resolves: #91 UserNote: When creating a bridge using the ARI the 'type' argument now accepts a new value 'sdp_label' which will configure the bridge to add labels for each stream in the SDP with the corresponding channel id.	2 years ago
InterLinked1	659f2aae3a	res_pjsip_pubsub: Add new pubsub module capabilities. (#82 ) The existing res_pjsip_pubsub APIs are somewhat limited in what they can do. This adds a few API extensions that make it possible for PJSIP pubsub modules to implement richer features than is currently possible. * Allow pubsub modules to get a handle to pjsip_rx_data on subscription * Allow pubsub modules to run a callback when a subscription is renewed * Allow pubsub modules to run a callback for outgoing NOTIFYs, with a handle to the tdata, so that modules can append their own headers to the NOTIFYs This change does not add any features directly, but makes possible several new features that will be added in future changes. Resolves: #81 ASTERISK-30485 #close Master-Only: True	2 years ago
Sean Bright	ae6b56e357	res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#47 ) The functionality we are interested in is present only in pjsip 2.13 and newer. Resolves: #45	3 years ago
zhengsh	c8ce2c705d	res_sorcery_memory_cache.c: Fix memory leak (#56 ) Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue. Resolves: #55 ASTERISK-30429	3 years ago
Henning Westerholt	1a7866b172	chan_pjsip: also return all codecs on empty re-INVITE for late offers (#59 ) We should also return all codecs on an re-INVITE without SDP for a call that used late offer (e.g. no SDP in the initial INVITE, SDP in the ACK). Bugfix for feature introduced in ASTERISK-30193 (https://issues.asterisk.org/jira/browse/ASTERISK-30193) Migration from previous gerrit change that was not merged.	3 years ago
Maximilian Fridrich	cacd98bb29	res_pjsip: mediasec: Add Security-Client headers after 401 (#49 ) When using mediasec, requests sent after a 401 must still contain the Security-Client header according to draft-dawes-sipcore-mediasec-parameter. Resolves: #48	3 years ago
Naveen Albert	0119f3ad48	res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters. The current STIR/SHAKEN signing process is inconsistent with the RFCs in a couple ways that can cause interoperability issues. RFC8225 specifies that the keys must be ordered lexicographically, but currently the fields are simply ordered according to the order in which they were added to the JSON object, which is not compliant with the RFC and can cause issues with some carriers. To fix this, we now leverage libjansson's ability to dump a JSON object sorted by key value, yielding the correct field ordering. Additionally, telephone numbers must have any leading + prefix removed and must not contain characters outside of 0-9, *, and # in order to comply with the RFCs. Numbers are now properly formatted as such. ASTERISK-30407 #close Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2	3 years ago
Henning Westerholt	58e88dec90	chan_pjsip: fix music on hold continues after INVITE with replaces In a three party scenario with INVITE with replaces, we need to unhold the call, otherwise one party continues to get music on hold, and the call is not properly bridged between them. ASTERISK-30428 Change-Id: I5675df11e739be5226b328f8828d4b8d81fbefb4	3 years ago
Mike Bradeen	e494a55467	res_pjsip_pubsub: subscription cleanup changes There are two main parts of the change associated with this commit. These are driven by the change in call order of pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject when an in-dialog SUBSCRIBE is received. First, the previous behavior was for pjproject to call pubsub_on_rx_refresh before calling pubsub_on_evsub_state when an in-dialog SUBSCRIBE was received that changes the subscription state. If that change was a termination due to a re-SUBSCRIBE with an expires of 0, we used to use the call to pubsub_on_rx_refresh to set the substate of the evsub to TERMINATE_PENDING before pjproject could call pubsub_on_evsub_state. This substate let pubsub_on_evsub_state know that the subscription TERMINATED event could be ignored as there was still a subsequent NOTIFY that needed to be generated and another call to pubsub_on_evsub_state to come with it. That NOTIFY was sent via serialized_pubsub_on_refresh_timeout which would see the TERMINATE_PENDING state and transition it to TERMINATE_IN_PROGRESS before triggering another call to pubsub_on_evsub_state (which now would clean up the evsub.) The new pjproject behavior is to call pubsub_on_evsub_state before pubsub_on_rx_refresh. This means we no longer can set the state to TERMINATE_PENDING to tell pubsub_on_evsub_state that it can ignore the first TERMINATED event. To handle this, we now look directly at the event type, method type and the expires value to determine whether we want to ignore the event or use it to trigger the evsub cleanup. Second, pjproject now expects the NOTIFY to actually be sent during pubsub_on_rx_refresh and avoids the protocol violation inherent in sending a NOTIFY before the SUBSCRIBE is acknowledged by caching the sent NOTIFY then sending it after responding to the SUBSCRIBE. This requires we send the NOTIFY using the non-serialized pubsub_on_refresh_timeout directly and let pjproject handle the protocol violation. ASTERISK-30469 Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7	3 years ago
Jaco Kroon	3d86701a86	res_calendar: output busy state as part of show calendar. Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	3 years ago
Sean Bright	acd1513111	ael: Regenerate lexers and parsers. Various changes to ensure that the lexers and parsers can be correctly generated when REBUILD_PARSERS is enabled. Some notes: * Because of the version of flex we are using to generate the lexers (2.5.35) some post-processing in the Makefile is still required. * The generated lexers do not contain the problematic C99 check that was being replaced by the call to sed in the respective Makefiles so it was removed. * Since these files are generated, they will include trailing whitespace in some places. This does not need to be corrected. Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e	3 years ago
Sean Bright	03e7bbbce9	Revert "pbx_ael: Global variables are not expanded." This reverts commit `c448dcd2f0`. Reason for revert: Behavior change that breaks existing dialplan. ASTERISK-30472 #close Change-Id: I20e44b4081d6ee0fe54cde44ac71dcf2d146f909	3 years ago
Sean Bright	76a2b2703f	res_agi: RECORD FILE plays 2 beeps. Sending the "RECORD FILE" command without the optional `offset_samples` argument can result in two beeps playing on the channel. This bug has been present since Asterisk 0.3.0 (2003-02-06). ASTERISK-30457 #close Change-Id: I95e88aa59378784d7f0eb648843f090e6723b787	3 years ago
Holger Hans Peter Freyther	92849c8c62	res_http_media_cache: Introduce options and customize Make the existing CURL parameters configurable and allow to specify the usable protocols, proxy and DNS timeout. ASTERISK-30340 Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b	3 years ago
George Joseph	25f7753f46	res_pjsip: Replace invalid UTF-8 sequences in callerid name * Added a new function ast_utf8_replace_invalid_chars() to utf8.c that copies a string replacing any invalid UTF-8 sequences with the Unicode specified U+FFFD replacement character. For example: "abc\xffdef" becomes "abc\uFFFDdef". Any UTF-8 compliant implementation will show that character as a � character. * Updated res_pjsip:set_id_from_hdr() to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. * Updated stasis_channels:ast_channel_publish_varset to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. ASTERISK-27830 Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8	3 years ago
cmaj	fd95998b56	res_phoneprov.c: Multihomed SERVER cache prevention Phones moving between subnets on multi-homed server have their initially connected interface IP cached in the SERVER variable, even when it is not specified in the configuration files. This prevents phones from obtaining the correct SERVER variable value when they move to another subnet. ASTERISK-30388 #close Reported-by: cmaj Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92	3 years ago
Mike Bradeen	b44ffd0565	res_pjsip: Prevent SEGV in pjsip_evsub_send_request contributed pjproject - patch to check sub->pending_notify in evsub.c:on_tsx_state before calling pjsip_evsub_send_request() res_pjsip_pubsub - change post pjsip 2.13 behavior to use pubsub_on_refresh_timeout to avoid the ao2_cleanup call on the sub_tree. This is is because the final NOTIFY send is no longer the last place the sub_tree is referenced. ASTERISK-30419 Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438	3 years ago
Mike Bradeen	62a64686e2	res_pjsip: Upgraded bundled pjsip to 2.13 Removed multiple patches. Code chages in res_pjsip_pubsub due to changes in evsub. Pjsip now calls on_evsub_state() before on_rx_refresh(), so the sub tree deletion that used to take place in on_evsub_state() now must take place in on_rx_refresh(). Additionally, pjsip now requires that you send the NOTIFY from within on_rx_refresh(), otherwise it will assert when going to send the 200 OK. The idea is that it will look for this NOTIFY and cache it until after sending the response in order to deal with the self-imposed message mis-order. Asterisk previously dealt with this by pushing the NOTIFY in on_rx_refresh(), but pjsip now forces us to use it's method. Changes were required to configure in order to detect which way pjsip handles this as the two are not compatible for the reasons mentioned above. A corresponding change in testsuite is required in order to deal with the small interal timing changes caused by moving the NOTIFY send. ASTERISK-30325 Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f	3 years ago
Sean Bright	c85fc1278f	doxygen: Fix doxygen errors. Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
Naveen Albert	d1bec3623e	res_pjsip_session: Add overlap_context option. Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	3 years ago
sungtae kim	1da489a434	res_stasis_snoop: Fix snoop crash Added NULL pointer check and channel lock to prevent resource release while the chanspy is processing. ASTERISK-29604 Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24	3 years ago
Sean Bright	c448dcd2f0	pbx_ael: Global variables are not expanded. Variable references within global variable assignments are now expanded rather than being included literally. ASTERISK-30406 #close Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d	3 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	3 years ago
George Joseph	91415a83d1	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	3 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	3 years ago
Igor Goncharovsky	410150235a	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	3 years ago
George Joseph	7dc8773178	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	3 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `e66c5da145`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	3 years ago
Holger Hans Peter Freyther	1c9f8ad7a6	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	3 years ago
George Joseph	82d5239bcb	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	3 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	3 years ago

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5430 Commits (644ad2f9cd82978abde0668ec7a56882d5484c29)