asterisk

Commit Graph

Author	SHA1	Message	Date
Mike Bradeen	933490b758	res_pjsip: update qualify_timeout documentation with DNS note The documentation on qualify_timeout does not explicitly state that the timeout includes any time required to perform any needed DNS queries on the endpoint. If the OPTIONS response is delayed due to the DNS query, it can still render an endpoint as Unreachable if the net time is enough for qualify_timeout to expire. Resolves: #352	2 years ago
Mike Bradeen	8c934fb7ed	res_speech_aeap: add aeap error handling res_speech_aeap previously did not register an error handler with aeap, so it was not notified of a disconnect. This resulted in SpeechBackground never exiting upon a websocket disconnect. Resolves: #303	2 years ago
Tinet-mucw	edc674a6ca	res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes. From the gdb information, ast_websocket_read reads a message successfully, then transport_read is called in the serializer. During execution of pjsip_transport_down, ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop. After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages. This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue. In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop. Resolves: asterisk#299	3 years ago
Vitezslav Novy	5179f1af24	res_rtp_asterisk: fix wrong counter management in ioqueue objects In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased which prevents unused ICE TURN threads from being removed. Resolves: #301	3 years ago
Sean Bright	3fafd7c0b7	res_stasis_recording.c: Save recording state when unmuted. Fixes #322	3 years ago
Mike Bradeen	b53e946b59	res_speech_aeap: check for null format on response * Fixed issue in res_speech_aeap when unable to provide an input format to check against.	3 years ago
Bastian Triller	0c0b99c5a1	res_pjsip_session: Send Session Interval too small response Handle session interval lower than endpoint's configured minimum timer when sending first answer. Timer setting is checked during this step and needs to handled appropriately. Before this change, no response was sent at all. After this change a response with 422 Session Interval too small is sent to UAC.	3 years ago
MikeNaso	b286d1cca2	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	3 years ago
Sean Bright	29eb4fe620	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	3 years ago
George Joseph	64b29be702	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.	3 years ago
Naveen Albert	8be503b663	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.	3 years ago
Maximilian Fridrich	8824b845c4	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.	3 years ago
Holger Hans Peter Freyther	a10de8bc72	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	3 years ago
Sean Bright	b5762cd54e	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.	3 years ago
Sean Bright	e120694711	res_geolocation: Ensure required 'location_info' is present. Fixes #189	3 years ago
zhengsh	25a766f49d	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	3 years ago
Mike Bradeen	9c889911ad	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	3 years ago
Sean Bright	4a1edb9a8c	res_pjsip_rfc3326: Prefer Q.850 cause code over SIP. Resolves: #116	3 years ago
Ben Ford	ad120e5d0b	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	3 years ago
Sean Bright	b7eae29fb9	pjsip_transport_events.c: Use %zu printf specifier for size_t. Partially resolves #143.	3 years ago
Sean Bright	97b901a29b	res_crypto.c: Gracefully handle potential key filename truncation. Partially resolves #143.	3 years ago
zhou_jiajian	f7e1f1fbbd	res_fax_spandsp.c: Clean up a spaces/tabs issue	3 years ago
Sean Bright	ebc007322b	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.	3 years ago
Jiajian Zhou	8e3f9dcc7b	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.	3 years ago
Naveen Albert	edf488c76e	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.	3 years ago
Joe Searle	ec64828c8f	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.	3 years ago
Sean Bright	2d18fb6e9d	res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77 ) The functionality we are interested in is present only in pjsip 2.13 and newer. Resolves: #45	3 years ago
zhengsh	8abb294063	res_sorcery_memory_cache.c: Fix memory leak 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	37a41a2a20	chan_pjsip: also return all codecs on empty re-INVITE for late offers 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	f3cc1e7fbd	res_pjsip: mediasec: Add Security-Client headers after 401 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	66a7cff57e	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	1c5720b802	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	df554a447c	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	019dc51139	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	34ff836db5	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	1ba3b34f77	Revert "pbx_ael: Global variables are not expanded." This reverts commit `56051d1ac5`. Reason for revert: Behavior change that breaks existing dialplan. ASTERISK-30472 #close Change-Id: I83bed3b800d36228a04ded0a6164b795f7f16bd6	3 years ago
Sean Bright	a3ec3efa02	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	8f088aa0f7	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	ceda5a9859	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	5b0e3444c3	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	37e558f6ef	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	58636a6ea6	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	96d9ad51ac	doxygen: Fix doxygen errors. Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
sungtae kim	f99849f8d5	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	56051d1ac5	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
Naveen Albert	a1da8042d1	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
George Joseph	2f5aece0c9	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
Igor Goncharovsky	3526441e41	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	4710f37ef6	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	62ca063fca	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `d454801c2d`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86	3 years ago

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5431 Commits (e9abf11a26ea34f5324757c9426ac52eaaf0462c)