730 Commits (914c8e28c1bb99b8217f93b51ce8f59870350fb6)

Author	SHA1	Message	Date
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
Henning Westerholt	093593a7c8	res_pjsip: return all codecs on a re-INVITE without SDP ... Currently chan_pjsip on receiving a re-INVITE without SDP will only return the codecs that are previously negotiated and not offering all enabled codecs. This causes interoperability issues with different equipment (e.g. from Cisco) for some of our customers and probably also in other scenarios involving 3PCC infrastructure. According to RFC 3261, section 14.2 we SHOULD return all codecs on a re-INVITE without SDP The PR proposes a new parameter to configure this behaviour: all_codecs_on_empty_reinvite. It includes the code, documentation, alembic migrations, CHANGES file and example configuration additions. ASTERISK-30193 #close Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148	3 years ago
Maximilian Fridrich	675bed5f2c	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	3 years ago
Maximilian Fridrich	479ce16eba	res_pjsip: Add 100rel option "peer_supported". ... This patch adds a new option to the 100rel parameter for pjsip endpoints called "peer_supported". When an endpoint with this option receives an incoming request and the request indicated support for the 100rel extension, then Asterisk will send 1xx responses reliably. If the request did not indicate 100rel support, Asterisk sends 1xx responses normally. ASTERISK-30158 Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad	3 years ago
Mike Bradeen	c49a2c46f6	alembic: add missing ps_endpoints columns ... The following required columns were missing, now added to the ps_endpoints table: incoming_call_offer_pref outgoing_call_offer_pref stir_shaken_profile ASTERISK-29453 Change-Id: I5cf565edf30195844d6acbc1e1de8c5f0d837568	3 years ago
George Joseph	cdbd973ecd	Geolocation: chan_pjsip Capability Preview ... This commit adds res_pjsip_geolocation which gives chan_pjsip the ability to use the core geolocation capabilities. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30128 Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9	3 years ago
Kevin Harwell	0ddbf6bc45	res_pjsip: allow TLS verification of wildcard cert-bearing servers ... Rightly the use of wildcards in certificates is disallowed in accordance with RFC5922. However, RFC2818 does make some allowances with regards to their use when using subject alt names with DNS name types. As such this patch creates a new setting for TLS transports called 'allow_wildcard_certs', which when it and 'verify_server' are both enabled allows DNS name types, as well as the common name that start with '*.' to match as a wildcard. For instance: *.example.com will match for: foo.example.com Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc... And the starting wildcard only matches for a single level. For instance: *.example.com will NOT match for: foo.bar.example.com The new setting is disabled by default. ASTERISK-30072 #close Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4	3 years ago
Naveen Albert	b81fbbc6dc	res_pjsip_outbound_registration: Make max random delay configurable. ... Currently, PJSIP will randomly wait up to 10 seconds for each outbound registration's initial attempt. The reason for this is to avoid having all outbound registrations attempt to register simultaneously. This can create limitations with the test suite where we need to be able to receive inbound calls potentially within 10 seconds of starting up. For instance, we might register to another server and then try to receive a call through the registration, but if the registration hasn't happened yet, this will fail, and hence this inconsistent behavior can cause tests to fail. Ultimately, this requires a smaller random value because there may be no good reason to wait for up to 10 seconds in these circumstances. To address this, a new config option is introduced which makes this maximum delay configurable. This allows, for instance, this to be set to a very small value in test systems to ensure that registrations happen immediately without an unnecessary delay, and can be used more generally to control how "tight" the initial outbound registrations are. ASTERISK-29965 #close Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd	3 years ago
Mark Petersen	16e59db514	chan_pjsip: add allow_sending_180_after_183 option ... added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	3 years ago
Alexei Gradinari	e2423c6f49	res_pjsip_pubsub: provide a display name for RLS subscriptions ... Whereas BLFs allow to show a display name for each RLS entry, the asterisk provides only the extension now. This is not end user friendly. This commit adds a new resource_list option, resource_display_name, to indicate whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. ASTERISK-29891 #close Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681	3 years ago
Frederic Van Espen	965f4abd9a	ast_coredumper: Fix deleting results when output dir is set ... When OUTPUTDIR is set to another directory and the --delete-results-after is set, the resulting txt files are not deleted. ASTERISK-29794 #close Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286	3 years ago
Alexander Traud	d914e14420	progdocs: Update Makefile. ... In developer mode, use internal documentation as well. This should produce no warnings. Fix yours! In noisy mode, output all possible warnings of Doxygen. This creates zillion of warnings. Double-check your current module! Any warnings are in the file './doxygen.log'. Beside that, this change avoids deprecated parameters because the configuration file for Doxygen contains only those parameters which differ from the default. This avoids the need to update the file on each run. Furthermore, it adds AST_VECTOR to be expanded. Finally, the default name for that file is Doxyfile. Therefore, let us use that! ASTERISK-26991 ASTERISK-20259 Change-Id: I4129092a199d5e24c319a09cd088614b121015af	3 years ago
Josh Soref	2a8b651b7e	contrib: Spelling fixes ... Correct typos of the following word families: standard increase comments valgrind promiscuous editing libtonezone storage aggressive whitespace russellbryant consecutive peternixon ASTERISK-29714 Change-Id: I9cafbf41b579c9c0c84c81719d2c4f900beec245	4 years ago
George Joseph	08cb67251f	ast_coredumper: Refactor to better find things ... The search for a running asterisk when --running is used has been greatly simplified and in the event it doesn't work, you can now specify a pid to use on the command line with --pid. The search for asterisk modules when --tarball-coredumps is used has been enhanced to have a better chance of finding them and in the event it doesn't work, you can now specify --libdir on the command line to indicate the library directory where they were installed. The DATEFORMAT variable was renamed to DATEOPTS and is now passed to the 'date' utility rather than running DATEFORMAT as a command. The coredump and output files are now renamed with DATEOPTS. This can be disabled by specifying --no-rename. Several confusing and conflicting options were removed: --append-coredumps --conffile --no-default-search --tarball-uniqueid The script was re-structured to make it easier for follow. Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59	4 years ago
Matthew Kern	15e432220c	res_pjsip_t38: bind UDPTL sessions like RTP ... In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the fallback use of the transport's bind address solve problems sending media on systems that cannot send ipv4 packets on ipv6 sockets, and certain other situations. This change extends both of these behaviors to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific problems on these systems, introducing a new option endpoint/t38_bind_udptl_to_media_address. ASTERISK-29402 Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557	4 years ago
Joseph Nadiv	4368764032	res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts ... The behavior of max_contacts and remove_existing are connected. If remove_existing is enabled, the soonest expiring contacts are removed. This may occur when there is an unavailable contact. Similarly, when remove_existing is not enabled, registrations from good endpoints are rejected in favor of retaining unavailable contacts. This commit adds a new AOR option remove_unavailable, and the effect of this setting will depend on remove_existing. If remove_existing is set to no, we will still remove unavailable contacts when they exceed max_contacts, if there are any. If remove_existing is set to yes, we will prioritize the removal of unavailable contacts before those that are expiring soonest. ASTERISK-29525 Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784	4 years ago
Naveen Albert	6873c5f3e4	sip_to_pjsip: Fix missing cases ... Adds the "auto" case which is valid with both chan_sip dtmfmode and chan_pjsip's dtmf_mode, adds subscribecontext to subscribe_context conversion, and accounts for cipher = ALL being invalid. ASTERISK-29459 Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2	4 years ago
Sean Bright	d2dcd15bd8	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	4 years ago
Joshua C. Colp	c2f4925ee0	svn: Switch to https scheme. ... Some versions of SVN seemingly don't follow the redirect to https. Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253	4 years ago
Sean Bright	35302efe73	app_queue: Add alembic migration to add ringinuse to queue_members. ... ASTERISK-28356 #close Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39	4 years ago
roadkill	cba8426b4c	contrib/systemd: Added note on common issues with systemd and asterisk ... With newer version of linux /var/run/ is a symlink to /run/ that has been turned into tmpfs. Added note that if asterisk has to bind to a specific IP that systemd has to wait until the network is up. Added note on how to make sure that the environment variable HOSTNAME is included. ASTERISK-29216 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: Ib3e560655befd3e99eec743687144f5569533379	4 years ago
Sean Bright	fe540d0326	sip_to_pjsip.py: Handle #include globs and other fixes ... * Wildcards in #includes are now properly expanded * Implement operators for Section class to allow sorting ASTERISK-29142 #close Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df	5 years ago
Alexander Traud	f89531cb98	install_prereq: Add GMime 3.0. ... Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not come with GMime 3.0. aptitude ignores any missing package. Therefore, it installs the correct package(s). However, in Ubuntu 18.04 LTS and Ubuntu 20.04 LTS, both versions are installed alongside although only one is really needed. Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7	5 years ago
Torrey Searle	83140c9fed	res_pjsip_diversion: implement support for History-Info ... Implemention of History-Info capable of interworking with Diversion Header following RFC7544 ASTERISK-29027 #close Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6	5 years ago
Sungtae Kim	1fd12b88c7	realtime: Increased reg_server character size ... Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20). This is fine for normal cases, but if the hostname is longer than 20, it returns error and then failed to register the contact address of the peer. Normally, 20 characters limitation for the hostname is fine, but with the cloud env. So, increased the size to 255. ASTERISK-29056 Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d	5 years ago
George Joseph	e32815dddb	ast_coredumper: Fix issues with naming ... If you run ast_coredumper --tarball-coredumps in the same directory as the actual coredump, tar can fail because the link to the actual coredump becomes recursive. The resulting tarball will have everything _except_ the coredump (which is usually what you need) There's also an issue that the directory name in the tarball is the same as the coredump so if you extract the tarball the directory it creates will overwrite the coredump. So: * Made the link to the coredump use the absolute path to the file instead of a relative one. This prevents the recursive link and allows tar to add the coredump. * The tarballed directory is now named <coredump>.output instead of just <coredump> so if you expand the tarball it won't overwrite the coredump. Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea	5 years ago
Alexander Traud	9ed1b1452d	sip_nat_settings: Update script for latest Linux. ... With the latest Linux, 'ifconfig' is not installed on default anymore. Furthermore, the output of the current net-tools 'ifconfig' changed. Therefore, parsing failed. This update uses 'ip addr show' instead. Finally, the service for the external IP changed. Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0	5 years ago
George Joseph	802aa97fa0	ACN: Configuration renaming for pjsip endpoint ... This change renames the codec preference endpoint options. incoming_offer_codec_prefs becomes codec_prefs_incoming_offer to keep the options together when showing an endpoint. Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d	5 years ago
Ben Ford	5fbed5af24	res_stir_shaken: Add stir_shaken option and general improvements. ... Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	5 years ago
George Joseph	2d22e34206	ACN: res_pjsip endpoint options ... This commit adds the endpoint options required to control Advanced Codec Negotiation. incoming_offer_codec_prefs outgoing_offer_codec_prefs incoming_answer_codec_prefs outgoing_answer_codec_prefs The documentation may need tweaking and some additional edits added, especially for the "answer" prefs. That'll be handled when things finalize. This commit is safe to merge as it doens't alter any existing functionality nor does it alter the previous codec negotiation work which may now be obsolete. Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11	5 years ago
sungtae kim	81b5e4a73f	res_pjsip.c: Added disable_rport option for pjsip.conf ... Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	5 years ago
Alexander Traud	a107e85b2e	install_prereq: Add libcap for high bits in DiffServ/ToS. ... works automatically; see Mantis 7047 (now ASTERISK-6863) Change-Id: I27d2c109180bd857b6757fd532de48eddb78aee6	5 years ago
Alexander Traud	610e058189	BuildSystem: Search for Python/C API when possibly needed only. ... The Python/C API is used only if the Test Framework was enabled in Asterisk 'make menuselect'. The Test Framework is available only if the Developer Mode was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API is used only if the PJProject was found and not disabled in Asterisk; the user did not go for './configure --without-pjproject'. Furthermore, because version 2 of that Python/C API is required (currently) and because some platforms do not offer a generic version 2, the script searches for 2.7 explicitly as well. To avoid version mismatch between the Python/C API and the Python environment, the script searches for the latter in the same versions, in the same the order as well. Because this Python/C API is just for (some) Asterisk contributors, the script also goes for the Python 3 environment as a last resort for all other Asterisk users. This allows 'make full' even on minimal installations of Ubuntu 18.04 LTS and newer. Because the Python/C API is Asterisk contributor specific, the Python packages are removed from the script './contrib/scripts/install_prereq' as this script is intended for Asterisk users. Asterisk contributors have to install much more packages in any case, like: sudo apt install autoconf automake git git-review python2.7-dev ASTERISK-28824 ASTERISK-27717 Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876	5 years ago
Kevin Harwell	26713dc88b	ast_coredumper: add Asterisk information dump ... This patch makes it so ast_coredumper now outputs the following information to a *-info.txt file when processing a core file: asterisk version and "built by" string BUILD_OPTS system start, and last reloaded date/time taskprocessor list equivalent of "bridge show all" equivalent of "core show channels verbose" Also a slight modification was made when trying to obtain the pid(s) of a running Asterisk. If it fails to retrieve any it now reports an error. Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b	5 years ago
Sean Bright	de6919f339	ast_tls_cert: Allow private key size to be set on command line ... The default size in release branches will be 1024 but we'll use 2048 in master. ASTERISK~28750 Change-Id: I435cea18bdd58824ed2b55259575c7ec7133842a	5 years ago
Sylvain Afchain	0c02d0a450	install_prereq: Install aptitude non-interactively ... Currently aptitude is installed using interactive mode. This patch changes this to use the non-interactive mode as it can block automatic dependencies installation, ex: CI, Docker build. ASTERISK-28726 #close Change-Id: I271ee00d230513a6f044810351a32d83b2181133	5 years ago
Joshua Colp	dbb3a19c35	Merge "queue_log: Add alembic script for generate db table for queue_log"	5 years ago
George Joseph	cc2d1f2545	Merge "contrib/valgrind: Fix use of frame-level suppression"	5 years ago
Snuffy	095c204fe0	contrib/valgrind: Fix use of frame-level suppression ... Fix use of frame-level wildcard usage in suppression file. ASTERISK-27243 #close Reported-by: Richard Kenner Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51	5 years ago
Pascal Cadotte Michaud	e494d5fd76	sip_to_pjsip.py: Fix trustrpid typo ... ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	5 years ago
Rodrigo Ramírez Norambuena	48161dfc71	queue_log: Add alembic script for generate db table for queue_log ... Change-Id: I35b928a6251f9da9a1742b2cd14c63a00c3d0f0c	6 years ago
Sean Bright	966488ab52	res_musiconhold: Add new 'playlist' mode ... Allow the list of files to be played to be provided explicitly in the music class's configuration. The primary driver for this change is to allow URLs to be used for MoH. Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa	6 years ago
Walter Doekes	4304c6534a	contrib/scripts: Make spandspflow2pcap.py Python 2.7+/3.3+ compatible ... Change-Id: Ica182a891743017ff3cda16de3d95335fffd9a91	6 years ago
Dan Cropp	cffa2a74cb	res_pjsip: Added a norefersub configuration setting ... Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	6 years ago
Ben Ford	4edd24841d	alembic: Fix errors during upgrade head. ... When trying to upgrade using alembic, a couple different errors kept popping up that prevented the upgrade. An additional parameter was needed when changing the schema for mwi_subscribe_replaces_unsolicited from an integer to an enum. When changing from a string to an enum, the type needed to be cast for postgresql. The other issue was a parameter being used during column creation that did not exist. After fixing the upgrade process, it revealed errors with the downgrade process. One was a variable not being defined in the downgrade function, and the other was tables not existing when using MySQL. This was due to a context check that should have encompassed MySQL, but in the end was not doing so. Change-Id: Ib4d70cf3ce5080023a50be496272a777b55d6c8e	6 years ago
cirillor	7d5409912f	Variable ALTCONF ignored when service is used in Debian ... When variable ALTCONF is defined, the command start prints the message "Unable to open specified master config file '"/etc/asterisk/asteris..." and use default configurations. ASTERISK-28332 Change-Id: I7595e582a0ee2c1051ea35435e247e27906957ef	6 years ago
Torrey Searle	4661c08549	chan_pjsip: add a flag to ignore 183 responses if no SDP present ... chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	6 years ago
Sean Bright	f098d4a325	sip_to_pjsip: Make multiline comment parsing consistent with Asterisk ... In Asterisk configuration, a multiline comment starts with ;-- as long as it is not followed by another dash (i.e. ;--- is not a multiline comment). ASTERISK-28323 #close Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72	6 years ago
George Joseph	c2adeb9dc2	taskprocessor: Enable subsystems and overload by subsystem ... To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	6 years ago
Alexei Gradinari	f0546d1d87	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	6 years ago