762 Commits (b4eb773d5ea9befa25e8045a57dba550a690e541)

Author	SHA1	Message	Date
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.	11 months ago
George Joseph	80839ce954	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	12 months ago
Sean Bright	07debc3b91	alembic: Correct NULLability of PJSIP id columns. ... Fixes #695	1 year 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	1 year 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	1 year 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	1 year 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.	1 year 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.	1 year ago
Sean Bright	1965574944	alembic: Quote new MySQL keyword 'qualify.' ... Fixes #651	1 year 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	1 year 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.	1 year 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	1 year 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	1 year 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	1 year 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.	1 year 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	2 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	2 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	2 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.	2 years ago
Sean Bright	fe15631d38	ast-db-manage: Fix alembic branching error caused by #122. ... Fixes #147.	2 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.	2 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.	2 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	2 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	2 years ago
The_Blode	de9aaf7e97	install_prereq: Add Linux Mint support. ... ASTERISK-30359 #close Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556	2 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	2 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	2 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	2 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	3 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	3 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	3 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	3 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	3 years ago
Mark Petersen	1cdaeb8161	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	c12cb899de	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	cb44ceadec	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	f3b29c6aa8	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	815e99d5ea	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	8aea2e5929	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	5e9799a42e	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	47cb177baf	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
Joshua C. Colp	20b2741232	chan_vpb: Remove deprecated module. ... ASTERISK-29597 Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1	4 years ago
Joshua C. Colp	6ecc48086c	chan_nbs: Remove deprecated module. ... ASTERISK-29595 Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33	4 years ago
Naveen Albert	c8bf8a54c2	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	4a843e00ef	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	4 years ago
Joshua C. Colp	531eb65cf3	svn: Switch to https scheme. ... Some versions of SVN seemingly don't follow the redirect to https. Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253	4 years ago