5305 Commits (79331255357666a2090df081870fbc145d7c3855)

Author	SHA1	Message	Date
Trevor Peirce	ff80a61118	res_pjsip: Actually enable session timers when timers=always ... When a pjsip endpoint is defined with timers=always, this has been a functional noop. This patch correctly sets the feature bitmap to both enable support for session timers and to enable them even when the endpoint itself does not request or support timers. ASTERISK-29603 Reported-By: Ray Crumrine Change-Id: I8b5eeaa9ec7f50cc6d96dd34c2b4aa9c53fb5440	3 years ago
Alexei Gradinari	1c5f023af9	res_pjsip_pubsub: delete scheduled notification on RLS update ... If there is scheduled notification, we must delete it to avoid using destroyed subscriptions. ASTERISK-29906 Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f	3 years ago
Alexei Gradinari	7480ebe9ea	res_pjsip_pubsub: XML sanitized RLS display name ... ASTERISK-29891 Change-Id: Ic8c9697e616446e06e6302653eae902aa23372ad	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
Naveen Albert	c277fd02e0	res_parking: Add music on hold override option. ... An m option to Park and ParkAndAnnounce now allows specifying a music on hold class override. ASTERISK-30087 Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e	3 years ago
Naveen Albert	6ae9a5835e	xmldocs: Improve examples. ... Use example tags instead of regular para tags where possible. ASTERISK-30090 Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5	3 years ago
Naveen Albert	d0d74d060b	res_calendar: Prevent assertion if event ends in past. ... res_calendar will trigger an assertion currently if the ending time is calculated to be in the past. Unlike the reminder and start times, however, there is currently no check to catch non-positive times and set them to 1. As a result, if we get a negative value by happenstance, this can cause a crash. To prevent the assertion from begin triggered, we now use the same logic as the reminder and start events to catch this issue before it can cause a problem. ASTERISK-29981 #close Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c	3 years ago
Naveen Albert	89f3def351	res_parking: Warn if out of bounds parking spot requested. ... Emits a warning if the user has requested a parking spot that is out of bounds for the requested parking lot. ASTERISK-30086 Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf	3 years ago
Alexei Gradinari	fa84b4c692	res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI ... The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local" Identity Element URI and Target Element URI to the same value - the channel Caller Number. For Identity Element it's ok to set as Caller ID. But Local Target URI should be set as local URI. In this case the Local Target URI can be used for Directed Call Pickup by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup). Also XML sanitized Display names. ASTERISK-24601 Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343	3 years ago
Shloime Rosenblum	1978732b8b	res_agi: Evaluate dialplan functions and variables in agi exec if enabled ... Agi commnad exec can now evaluate dialplan functions and variables if variable AGIEXECFULL is set to yes. this can be useful when executing Playback or Read from agi. ASTERISK-30058 #close Change-Id: I669991f540496e7bddd096fec82b52c083036832	3 years ago
Moritz Fain	aaa14d3c7d	ari: expose channel driver's unique id to ARI channel resource ... This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Joshua C. Colp	c5c858287a	res_pjsip_transport_websocket: Also set the remote name. ... As part of PJSIP 2.11 a behavior change was done to require a matching remote hostname on an established transport for secure transports. Since the Websocket transport is considered a secure transport this caused the existing connection to not be found and used. We now set the remote hostname and the transport can be found. ASTERISK-30065 Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94	4 years ago
Thomas Guebels	9f6bda1821	res_pjsip_transport_websocket: save the original contact host ... This is needed to be able to restore it in REGISTER responses, otherwise the client won't be able to find the contact it created. ASTERISK-30042 Change-Id: I0c5823918199acf09246b3b206fbde66773688f6	4 years ago
Naveen Albert	d5d3788648	res_pjsip_outbound_registration: Show time until expiration ... Adjusts the pjsip show registration(s) commands to show the amount of seconds remaining until a registration expires. ASTERISK-29845 #close Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483	4 years ago
George Joseph	ad6af63895	GCC12: Fixes for 16+ ... Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570	4 years ago
Naveen Albert	4aac359d79	documentation: Adds versioning information. ... Adds version information for applications, functions, and manager events/actions. This is not completely exhaustive by any means but covers most new things added that have release versioning information in the issue tracker. ASTERISK-29940 #close Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131	4 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	4 years ago
Kevin Harwell	2fb8667908	res_aeap & res_speech_aeap: Add Asterisk External Application Protocol ... Add framework to connect to, and read and write protocol based messages from and to an external application using an Asterisk External Application Protocol (AEAP). This has been divided into several abstractions: 1. transport - base communication layer (currently websocket only) 2. message - AEAP description and data (currently JSON only) 3. transaction - links/binds requests and responses 4. aeap - transport, message, and transaction handler/manager This patch also adds an AEAP implementation for speech to text. Existing speech API callbacks for speech to text have been completed making it possible for Asterisk to connect to a configured external translator service and provide audio for STT. Results can also be received from the external translator, and made available as speech results in Asterisk. Unit tests have also been created that test the AEAP framework, and also the speech to text implementation. ASTERISK-29726 #close Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2	4 years ago
Ben Ford	62f8e157fb	res_aeap: Add basic config skeleton and CLI commands. ... Added support for a basic AEAP configuration read from aeap.conf. Also added 2 CLI commands for showing individual configurations as well as all of them: aeap show server <id> and aeap show servers. Only one configuration option is required at the moment, and that one is server_url. It must be a websocket URL. The other option, codecs, is optional and will be used over the codecs specified on the endpoint if provided. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453 Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3 (cherry picked from commit 45a1977de4)	4 years ago
Joshua C. Colp	09e8667fa5	res_pjsip: Always set async_operations to 1. ... The async_operations setting on a transport configures how many simultaneous incoming packets the transport can handle when multiple threads are polling and waiting on the transport. As we only use a single thread this was needlessly creating incoming packets when set to a non-default value, wasting memory. ASTERISK-30006 Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f	4 years ago
Ben Ford	40f4268f2d	res_pjsip_stir_shaken.c: Fix enabled when not configured. ... There was an issue with the conditional where STIR/SHAKEN would be enabled even when not configured. It has been changed to ensure that if a profile does not exist and stir_shaken is not set in pjsip.conf, then the conditional will return from the function without performing STIR/SHAKEN operations. ASTERISK-30024 Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6	4 years ago
Ben Ford	11accf8064	AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. ... Adds a new configuration option, stir_shaken_profile, in pjsip.conf that can be specified on a per endpoint basis. This option will reference a stir_shaken_profile that can be configured in stir_shaken.conf. The type of this option must be 'profile'. The stir_shaken option can be specified on this object with the same values as before (attest, verify, on), but it cannot be off since having the profile itself implies wanting STIR/SHAKEN support. You can also specify an ACL from acl.conf (along with permit and deny lines in the object itself) that will be used to limit what interfaces Asterisk will attempt to retrieve information from when reading the Identity header. ASTERISK-29476 Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406	4 years ago
Ben Ford	33091c2659	AST-2022-001 - res_stir_shaken/curl: Limit file size and check start. ... Put checks in place to limit how much we will actually download, as well as a check for the data we receive at the start to ensure it begins with what we would expect a certificate to begin with. ASTERISK-29872 Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46	4 years ago
Boris P. Korzun	82dbfe7783	res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity ... Change RTP timer behavior for detecting RTP only after two-way SDP channel establishment. Ignore detecting after receiving 183 with SDP or while direct media is used. Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout and rtpholdtimeout options in chan_sip. ASTERISK-26689 #close ASTERISK-29929 #close Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb	4 years ago
Kevin Harwell	ec5b449bcf	res_pjsip_header_funcs: wrong pool used tdata headers ... When adding headers to an outgoing request the headers were cloned using the dialog's pool when they should have been cloned using tdata's pool. Under certain circumstances it was possible for the dialog object, and its pool to be freed while tdata is still active and available. Thus the cloned header "disappeared", and when tdata tried to later access it a crash would occur. This patch makes it so all added headers are cloned appropriately using tdata's pool. ASTERISK-29411 #close ASTERISK-29535 #close Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5	4 years ago
Sean Bright	777e9fde67	openssl: Supress deprecation warnings from OpenSSL 3.0 ... There is work going on to update our OpenSSL usage to avoid the deprecated functions but in the meantime make it possible to compile in devmode. Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726	4 years ago
Naveen Albert	dc129b6951	res_agi: Fix xmldocs bug with set music. ... The XML documentation for the SET MUSIC AGI command is invalid, as the parameter does not have a name and the on/off enum options for the on/off argument are listed separately, which is incorrect. The cumulative effect of these currently is that the Asterisk Wiki documentation for SET MUSIC is broken and external documentation generators crash on SET MUSIC due to the malformed documentation. These issues are corrected so that the documentation can be successfully parsed as with other similar AGI commands. ASTERISK-29939 #close ASTERISK-28891 #close Change-Id: I8c3d59897531bcbc401cbc7b00c9e2829dcb35f8 (cherry picked from commit 37ece75677)	4 years ago
Philip Prindeville	f50e793665	time: add support for time64 libcs ... Treat time_t's as entirely unique and use the POSIX API's for converting to/from strings. Lastly, a 64-bit integer formats as 20 digits at most in base10. Don't need to have any 100 byte buffers to hold that. ASTERISK-29674 #close Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com> Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56	4 years ago
Alexei Gradinari	96a3ff9edd	res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request ... When asterisk generates the RLMI part of NOTIFY request, the asterisk uses the local contact uri instead of the URI to which the SUBSCRIBE request is sent. Because of this mismatch some IP phones (for example Cisco 5XX) ignore this list. According https://datatracker.ietf.org/doc/html/rfc4662#section-5.2 The first mandatory <list> attribute is "uri", which contains the uri that corresponds to the list. Typically, this is the URI to which the SUBSCRIBE request was sent. https://datatracker.ietf.org/doc/html/rfc4662#section-5.3 The "uri" attribute identifies the resource to which the <resource> element corresponds. Typically, this will be a SIP URI that, if subscribed to, would return the state of the resource. This patch makes asterisk to generate URI using SUBSCRIBE request URI. ASTERISK-29961 #close Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c	4 years ago
Sean Bright	3a7d83087b	stasis_recording: Perform a complete match on requested filename. ... Using the length of a file found on the filesystem rather than the file being requested could result in filenames whose names are substrings of another to be erroneously matched. We now ensure a complete comparison before returning a positive result. ASTERISK-29960 #close Change-Id: Id3ffc77681b9b75b8569062f3d952a128a21c71a	4 years ago
Sean Bright	2b636f3766	download_externals: Use HTTPS for downloads ... ASTERISK-29980 #close Change-Id: I7b347665822ea2774dd322276c09be67914d2065	4 years ago
Boris P. Korzun	5b653b8a7b	res_config_pgsql: Add text-type column check in require_pgsql() ... Omit "unsupported column type 'text'" warning in logs while using text-type column in the PgSQL backend. ASTERISK-29924 #close Change-Id: I48061a7d469426859670db07f1ed8af1eb814712	4 years ago
Alexei Gradinari	8666455bd8	res_pjsip_pubsub: update RLS to reflect the changes to the lists ... This patch makes the Resource List Subscriptions (RLS) dynamic. The asterisk updates the current subscriptions to reflect the changes to the list on the subscriptions refresh. If list items are added, removed, updated or do not exist anymore, the asterisk regenerates the resource list. ASTERISK-29906 #close Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3	4 years ago
George Joseph	468441121d	xmldoc: Fix issue with xmlstarlet validation ... Added the missing xml-stylesheet and Xinclude namespace declarations in pjsip_config.xml and pjsip_manager.xml. Updated make_xml_documentation to show detailed errors when xmlstarlet is the validator. It's now run once with the '-q' option to suppress harmless/expected messages and if it actually fails, it's run again without '-q' but with '-e' to show the actual errors. Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e	4 years ago
George Joseph	a81e14d2da	Makefile: Allow XML documentation to exist outside source files ... Moved the xmldoc build logic from the top-level Makefile into its own script "make_xml_documentation" in the build_tools directory. Created a new utility script "get_sourceable_makeopts", also in the build_tools directory, that dumps the top-level "makeopts" file in a format that can be "sourced" from shell sscripts. This allows scripts to easily get the values of common make build variables such as the location of the GREP, SED, AWK, etc. utilities as well as the AST* and library *_LIB and *_INCLUDE variables. Besides moving logic out of the Makefile, some optimizations were done like removing "third-party" from the list of subdirectories to be searched for documentation and changing some assignments from "=" to ":=" so they're only evaluated once. The speed increase is noticeable. The makeopts.in file was updated to include the paths to REALPATH and DIRNAME. The ./conifgure script was setting them but makeopts.in wasn't including them. So... With this change, you can now place documentation in any"c" source file AND you can now place it in a separate XML file altogether. The following are examples of valid locations: res/res_pjsip.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_configuration.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_doc.xml A fully-formed XML file. The "configInfo", "manager", "managerEvent", etc. elements that would be in the "c" file DOCUMENTATION fragment should be wrapped in proper XML. Example for "somemodule.xml": <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE docs SYSTEM "appdocsxml.dtd"> <docs> <configInfo> ... </configInfo> </docs> It's the "appdocsxml.dtd" that tells make_xml_documentation that this is a documentation XML file and not some other XML file. It also allows many XML-capable editors to do formatting and validation. Other than the ".xml" suffix, the name of the file is not significant. As a start... This change also moves the documentation that was in res_pjsip.c to 2 new XML files in res/res_pjsip: pjsip_config.xml and pjsip_manager.xml. This cut the number of lines in res_pjsip.c in half. :) Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade	4 years ago
Naveen Albert	63db7505f2	asterisk: Add macro for curl user agent. ... Currently, each module that uses libcurl duplicates the standard Asterisk curl user agent. This adds a global macro for the Asterisk user agent used for curl requests to eliminate this duplication. ASTERISK-29861 #close Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d	4 years ago
Naveen Albert	74742cdb5e	res_stir_shaken: refactor utility function ... Refactors temp file utility function into file.c. ASTERISK-29809 #close Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b	4 years ago
Alexei Gradinari	1cc1fb54e7	res_pjsip_pubsub: fix Batched Notifications stop working ... If Subscription refresh occurred between when the batched notification was scheduled and the serialized notification was to be sent, then new schedule notification task would never be added. There are 2 threads: thread #1. ast_sip_subscription_notify is called, if notification_batch_interval then call schedule_notification. 1.1. The schedule_notification checks notify_sched_id > -1 not true, then send_scheduled_notify = 1 notify_sched_id = ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb.... 1.2. The sched_cb pushes task serialized_send_notify to serializer and returns 0 which means no reschedule. 1.3. The serialized_send_notify checks send_scheduled_notify if it's false the just returns. BUT notify_sched_id is still set, so no more ast_sched_add. thread #2. pubsub_on_rx_refresh is called 2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer 2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout which calls send_notify 2.3. The send_notify set send_scheduled_notify = 0; The serialized_send_notify should always unset notify_sched_id. ASTERISK-29904 #close Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875	4 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	4 years ago
Naveen Albert	74e9b60bd0	documentation: Adds missing default attributes. ... The configObject tag contains a default attribute which allows the default value to be specified, if applicable. This allows for the default value to show up specially on the wiki in a way that is clear to users. There are a couple places in the tree where default values are included in the description as opposed to as attributes, which means these can't be parsed specially for the wiki. These are changed to use the attribute instead of being included in the text description. ASTERISK-29898 #close Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755	4 years ago
Mark Petersen	6659e502a4	res_prometheus.c: missing module dependency ... added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO which fixes issue with module crashes on load "FRACK!, Failed assertion" ASTERISK-29871 Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096	4 years ago
Sean Bright	de0c29de55	res_pjsip.c: Correct minor typos in 'realm' documentation. ... Change-Id: I886936b808def5540d40071321e72f6bfa19063a	4 years ago
George Joseph	2a34bb1e11	res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup ... In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess that hasn't been initialized, it'll assert and abort. If digest_create_request_with_auth() fails to find the proper auth object however, it jumps to its cleanup which does exactly that. So now we no longer attempt to call pjsip_auth_clt_deinit() if we never actually initialized it. ASTERISK-29888 Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62	4 years ago
Naveen Albert	a4b01ececb	res_tonedetect: Fixes some logic issues and typos ... Fixes some minor logic issues with the module: Previously, the OPT_END_FILTER flag was getting tested before options were parsed, so it could never evaluate to true (wrong ordering). Additionally, the initially parsed timeout (float) needs to be compared with 0, not the result int which is set afterwards (wrong variable). ASTERISK-29857 #close Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c	4 years ago
Torrey Searle	9c9083b45a	res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf ... When generating dtmfs, asterisk can incorrectly think packet loss occured during the dtmf generation, resulting in a jump in sequence numbers when forwarding voice frames resumes. This patch forces asterisk to re-learn the expected sequence number after each DTMF to avoid this ASTERISK-29869 #close Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853	4 years ago
Kevin Harwell	98f86697cc	res_http_websocket: Add a client connection timeout ... Previously there was no way to specify a connection timeout when attempting to connect a websocket client to a server. This patch makes it possible to now do such. Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d	4 years ago
Luke Escude	6e8bbe4b3a	res_pjsip_sdp_rtp.c: Support keepalive for video streams. ... ASTERISK-28890 #close Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1	4 years ago
Naveen Albert	a9e9e15c3a	res_rtp_asterisk: Fix typo in flag test/set ... The code currently checks to see if an RFC3389 warning flag is set, except if it is, it merely sets the flag again, the logic of which doesn't make any sense. This adjusts the if comparison to check if the flag has NOT been set, and if so, emit a notice log event and set the flag so that future frames do not cause an event to be logged. ASTERISK-29856 #close Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1	4 years ago
George Joseph	f55886a72c	res_pjsip: Make message_filter and session multipart aware ... Neither pjsip_message_filter's filter_on_tx_message() nor res_pjsip_session's session_outgoing_nat_hook() were multipart aware and just assumed that an SDP would be the only thing in a message body. Both were changed to use the new pjsip_get_sdp_info() function which searches for an sdp in both single- and multi- part message bodies. ASTERISK-29813 Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56	4 years ago
George Joseph	59cf9f0047	res_pjsip: Add utils for checking media types ... Added two new functions to assist checking media types... * ast_sip_are_media_types_equal compares two pjsip_media_types. * ast_sip_is_media_type_in tests if one media type is in a list of others. Added static definitions for commonly used media types to res_pjsip.h. Changed several modules to use the new functions and static definitions. ASTERISK_29813 (not ready to close) Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9	4 years ago