775 Commits (a5d37e5ecba18d56dd604935ea69ef696e43ba31)

Author	SHA1	Message	Date
George Joseph	8706a485c3	alembic: Database updates required. ... This commit doesn't actually change anything. It just adds the following upgrade notes that were omitted from the original commits. Resolves: #1097 UpgradeNote: Two commits in this release... 'Add SHA-256 and SHA-512-256 as authentication digest algorithms' 'res_pjsip: Add new AOR option "qualify_2xx_only"' ...have modified alembic scripts for the following database tables: ps_aors, ps_contacts, ps_auths, ps_globals. If you don't use the scripts to update your database, reads from those tables will succeeed but inserts into the ps_contacts table by res_pjsip_registrar will fail.	1 year ago
Naveen Albert	0bfbabee41	ast_tls_cert: Add option to skip passphrase for CA private key. ... Currently, the ast_tls_cert file is hardcoded to use the -des3 option for 3DES encryption, and the script needs to be manually modified to not require a passphrase. Add an option (-e) that disables encryption of the CA private key so no passphrase is required. Resolves: #1064	1 year ago
George Joseph	a0987672f0	Add SHA-256 and SHA-512-256 as authentication digest algorithms ... * Refactored pjproject code to support the new algorithms and added a patch file to third-party/pjproject/patches * Added new parameters to the pjsip auth object: * password_digest = <algorithm>:<digest> * supported_algorithms_uac = List of algorithms to support when acting as a UAC. * supported_algorithms_uas = List of algorithms to support when acting as a UAS. See the auth object in pjsip.conf.sample for detailed info. * Updated both res_pjsip_authenticator_digest.c (for UAS) and res_pjsip_outbound_authentocator_digest.c (UAC) to suport the new algorithms. The new algorithms are only available with the bundled version of pjproject, or an external version > 2.14.1. OpenSSL version 1.1.1 or greater is required to support SHA-512-256. Resolves: #948 UserNote: The SHA-256 and SHA-512-256 algorithms are now available for authentication as both a UAS and a UAC.	1 year ago
Abdelkader Boudih	dd592b195b	normalize contrib/ast-db-manage/queue_log.ini.sample	1 year ago
George Joseph	c193752ca4	Add C++ Standard detection to configure and fix a new C++20 compile issue ... * The autoconf-archive package contains macros useful for detecting C++ standard and testing other C++ capabilities but that package was never included in the install_prereq script so many existing build environments won't have it. Even if it is installed, older versions won't newer C++ standards and will actually cause an error if you try to test for that version. To make it available for those environments, the ax_cxx_compile_stdcxx.m4 macro has copied from the latest release of autoconf-archive into the autoconf directory. * A convenience wrapper(ast_cxx_check_std) around ax_cxx_compile_stdcxx was also added so checking the standard version and setting the asterisk-specific PBX_ variables becomes a one-liner: `AST_CXX_CHECK_STD([std], [force_latest_std])`. Calling that with a version of `17` for instance, will set PBX_CXX17 to 0 or 1 depending on whether the current c++ compiler supports stdc++17. HAVE_CXX17 will also be 'defined" or not depending on the result. * C++ compilers hardly ever default to the latest standard they support. g++ version 14 for instance supports up to C++23 but only uses C++17 by default. If you want to use C++23, you have to add `-std=gnu++=23` to the g++ command line. If you set the second argument of AST_CXX_CHECK_STD to "yes", the macro will automatically keep the highest `-std=gnu++` value that worked and pass that to the Makefiles. * The autoconf-archive package was added to install_prereq for future use. * Updated configure.ac to use AST_CXX_CHECK_STD() to check for C++ versions 11, 14, 17, 20 and 23. * Updated configure.ac to accept the `--enable-latest-cxx-std` option which will set the second option to AST_CXX_CHECK_STD() to "yes". The default is "no". * ast_copy_string() in strings.h declares the 'sz' variable as volatile and does an `sz--` on it later. C++20 no longer allows the `++` and `--` increment and decrement operators to be used on variables declared as volatile however so that was changed to `sz -= 1`.	1 year ago
Kent	0b30f546ba	res_pjsip: Add new AOR option "qualify_2xx_only" ... Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify feature to mark a contact as available only if an OPTIONS request returns a 2XX response. If the option is not specified or is false, any response to the OPTIONS request marks the contact as available. UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only" option that can be set so that only 2XX responses to OPTIONS requests used to qualify a contact will mark the contact as available.	1 year ago
George Joseph	85a9bce879	res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T ... The suppress_moh_on_sendonly endpoint option should have been defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES in the alembic script instead of OPT_YESNO_T and YESNO_VALUES. Also updated contrib/ast-db-manage/README.md to indicate that AST_BOOL_VALUES should always be used and provided an example. Resolves: #995	1 year ago
George Joseph	98510d4c75	res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" ... Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party. Resolves: #979 UserNote: The new "suppress_moh_on_sendonly" endpoint option can be used to prevent playing MOH back to a caller if the remote end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.	1 year ago
Sean Bright	d9c14cba49	alembic: Drop redundant voicemail_messages index. ... The `voicemail_messages_dir` index is a left prefix of the table's primary key and therefore unnecessary.	1 year ago
Sean Bright	147de579a5	res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery ... Fixes #895	1 year ago
Sean Bright	65fee5f825	alembic: Make 'revises' header comment match reality.	2 years ago
Ben Ford	7990f6b589	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	2 years ago
George Joseph	59d75aea15	ast-db-manage: Remove duplicate enum creation ... Remove duplicate creation of ast_bool_values from 2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was causing alembic upgrades to fail since the enum was already created in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018. Resolves: #797	2 years ago
Alexei Gradinari	41d4db99cf	app_queue: Add option to not log Restricted Caller ID to queue_log ... Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event in the queue log if the Caller ID is restricted. Resolves: #765 UpgradeNote: Add a new column to the queues table: queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') to control whether the Restricted Caller ID will be stored in the queue log. UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID will be stored in the queue log. If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.	2 years ago
George Joseph	80839ce954	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	2 years ago
Sean Bright	07debc3b91	alembic: Correct NULLability of PJSIP id columns. ... Fixes #695	2 years ago
Sperl Viktor	7c3e2cfda9	res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address. ... Add a new identify_by option to res_pjsip_endpoint_identifier_ip called 'transport' this matches endpoints based on the bound ip address (local) instead of the 'ip' option, which matches on the source ip address (remote). UserNote: set identify_by=transport for the pjsip endpoint. Then use the existing 'match' option and the new 'transport' option of the identify. Fixes: #672	2 years ago
Sperl Viktor	136bd5c892	res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI ... Add ability to match against PJSIP request URI. UserNote: this new feature let users match endpoints based on the indound SIP requests' URI. To do so, add 'request_uri' to the endpoint's 'identify_by' option. The 'match_request_uri' option of the identify can be an exact match for the entire request uri, or a regular expression (between slashes). It's quite similar to the header identifer. Fixes: #599	2 years ago
Joshua Elson	3d40d34271	Implement Configurable TCP Keepalive Settings in PJSIP Transports ... This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary. Fixes #657	2 years ago
Sean Bright	8715a700e2	alembic: Fix compatibility with SQLAlchemy 2.0+. ... SQLAlchemy 2.0 changed the way that commits/rollbacks are handled causing the final `UPDATE` to our `alembic_version_<whatever>` tables to be rolled back instead of committed. We now use one connection to determine which `alembic_version_<whatever>` table to use and another to run the actual migrations. This prevents the erroneous rollback. This change is compatible with both SQLAlchemy 1.4 and 2.0.	2 years ago
Sean Bright	4ecbac98d8	res_pjsip: Fix alembic downgrade for boolean columns. ... When downgrading, ensure that we don't touch columns that didn't actually change during upgrade.	2 years ago
Sean Bright	1965574944	alembic: Quote new MySQL keyword 'qualify.' ... Fixes #651	2 years ago
Sean Bright	93e40968c3	res_pjsip: Use consistent type for boolean columns. ... This migrates the relevant schema objects from the `('yes', 'no')` definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')` one. Fixes #617	2 years ago
George Joseph	e6c7f1aee0	Stir/Shaken Refactor ... Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.	2 years ago
Sean Bright	d122225279	alembic: Synchronize alembic heads between supported branches. ... This adds a dummy migration to 18 and 20 so that our alembic heads are synchronized across all supported branches. In this case the migration we are stubbing (24c12d8e9014) is: 775352ee6c	2 years ago
Mike Bradeen	f5d622413a	app_voicemail: add NoOp alembic script to maintain sync ... Adding a NoOp alembic script for the voicemail database to maintain version sync with other branches. Fixes: #527	2 years ago
George Joseph	44f1522907	ast_coredumper: Increase reliability ... Instead of searching for the asterisk binary and the modules in the filesystem, we now get their locations, along with libdir, from the coredump itself... For the binary, we can use `gdb -c <coredump> ... "info proc exe"`. gdb can print this even without having the executable and symbols. Once we have the binary, we can get the location of the modules with `gdb ... "print ast_config_AST_MODULE_DIR` If there was no result then either it's not an asterisk coredump or there were no symbols loaded. Either way, it's not usable. For libdir, we now run "strings" on the note0 section of the coredump (which has the shared library -> memory address xref) and search for "libasteriskssl|libasteriskpj", then take the dirname. Since we're now getting everything from the coredump, it has to be correct as long as we're not crossing namespace boundaries like running asterisk in a docker container but trying to run ast_coredumper from the host using a shared file system (which you shouldn't be doing). There is still a case for using --asterisk-bin and/or --libdir: If you've updated asterisk since the coredump was taken, the binary, libraries and modules won't match the coredump which will render it useless. If you can restore or rebuild the original files that match the coredump and place them in a temporary directory, you can use --asterisk-bin, --libdir, and a new --moddir option to point to them and they'll be correctly captured in a tarball created with --tarball-coredumps. If you also use --tarball-config, you can use a new --etcdir option to point to what normally would be the /etc/asterisk directory. Also addressed many "shellcheck" findings. Resolves: #445	2 years ago
Sean Bright	0dcf03e844	alembic: Update list of TLS methods available on ps_transports. ... Related to #221 and #222. Also adds `*.ini` to the `.gitignore` file in ast-db-manage for convenience.	2 years ago
Sean Bright	978d09fc35	live_ast: Add astcachedir to generated asterisk.conf. ... `astcachedir` (added in b0842713) was not added to `live_ast` so continued to point to the system `/var/cache` directory instead of the one in the live environment.	2 years ago
sungtae kim	f89e56c178	res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters ... This commit introduces an extension to the endpoint and relevant resource sizes for PJSIP, transitioning from its current 40-character constraint to a more versatile 255-character capacity. This enhancement significantly overcomes limitations related to domain qualification and practical usage, ultimately delivering improved functionality. In addition, it includes adjustments to accommodate the expanded realm size within the ARI, specifically enhancing the maximum realm length. Resolves: #345 UserNote: With this update, the PJSIP realm lengths have been extended to support up to 255 characters. UpgradeNote: As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas.	2 years ago
George Joseph	c929146c61	safe_asterisk: Change directory permissions to 755 ... If the safe_asterisk script detects that the /var/lib/asterisk directory doesn't exist, it now creates it with 755 permissions instead of 770. safe_asterisk needing to create that directory should be extremely rare though because it's normally created by 'make install' which already sets the permissions to 755. Resolves: #316	3 years ago
George Joseph	cb9223cdb9	alembic: Fix quoting of the 100rel column ... Add quoting around the ps_endpoints 100rel column in the ALTER statements. Although alembic doesn't complain when generating sql statements, postgresql does (rightly so). Resolves: #274	3 years ago
Jason D. McCormick	fc57dca50d	install_prereq: Fix dependency install on aarch64. ... Fixes dependency solutions in install_prereq for Debian aarch64 platforms. install_prereq was attempting to forcibly install 32-bit armhf packages due to the aptitude search for dependencies. Resolves: #37	3 years ago
Sean Bright	ea63148b23	ast-db-manage: Synchronize revisions between comments and code. ... In a handful of migrations, the comment header that indicates the current and previous revisions has drifted from the identifiers revision and down_revision variables. This updates the comment headers to match the code.	3 years ago
Sean Bright	fe15631d38	ast-db-manage: Fix alembic branching error caused by #122. ... Fixes #147.	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
Niklas Larsson	df774619fb	app_queue: Preserve reason for realtime queues ... When Asterisk is restarted it does not preserve paused reason for members of realtime queues. This was fixed for non-realtime queues in ASTERISK_25732 Resolves: #66 UpgradeNote: Add a new column to the queue_member table: reason_paused VARCHAR(80) so the reason can be preserved. UserNote: Make paused reason in realtime queues persist an Asterisk restart. This was fixed for non-realtime queues in ASTERISK_25732.	3 years ago
Naveen Albert	22c9d52289	logrotate: Fix duplicate log entries. ... The Asterisk logrotate script contains explicit references to files with the .log extension, which are also included when *log is expanded. This causes issues with newer versions of logrotate. This fixes this by ensuring that a log file cannot be referenced multiple times after expansion occurs. Resolves: #96 ASTERISK-30442 Reported by: EN Barnett Tested by: EN Barnett	3 years ago
Sean Bright	5c6d5ea38f	core: Cleanup gerrit and JIRA references. (#57) ... * Remove .gitreview and switch to pulling the main asterisk branch version from configure.ac instead. * Replace references to JIRA with GitHub. * Other minor cleanup found along the way. Resolves: #39	3 years ago
The_Blode	de9aaf7e97	install_prereq: Add Linux Mint support. ... ASTERISK-30359 #close Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556	3 years ago
Sean Bright	46bdd5e3be	contrib: rc.archlinux.asterisk uses invalid redirect. ... `rc.archlinux.asterisk`, which explicitly requests bash in its shebang, uses the following command syntax: ${DAEMON} -rx "core stop now" > /dev/null 2&>1 The intent of which is to execute: ${DAEMON} -rx "core stop now" While sending both stdout and stderr to `/dev/null`. Unfortunately, because the `&` is in the wrong place, bash is interpreting the `2` as just an additional argument to the `$DAEMON` command and not as a file descriptor and proceeds to use the bashism `&>` to send stderr and stdout to a file named `1`. So we clean it up and just use bash's shortcut syntax. Issue raised and a fix suggested (but not used) by peutch on GitHub¹. ASTERISK-30449 #close 1. https://github.com/asterisk/asterisk/pull/31 Change-Id: Ie279bf4efb4d95cbf507313483d316e977303d19	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
Michael Kuron	fee9012fe1	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	3 years ago
Henning Westerholt	12445040d3	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	0d2e140123	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	5bbad0d27c	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	4 years ago
Mike Bradeen	46776c77c4	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	4 years ago
George Joseph	1fa568e76f	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	4 years ago
Kevin Harwell	a3b2daf127	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	4 years ago
Naveen Albert	31dc28ab09	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	4 years ago