493 Commits (5bfe94461a96baa60399a3266a268b1e727d5c3f)

Author	SHA1	Message	Date
George Joseph	dd76b53ca6	pjsip_transport_events: Fix possible use after free on transport ... It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	3 years ago
George Joseph	5e3a2dd5b0	pjsip_config.xml: Fix omissions caused by wrong cherry-pick order ... When cherry-picking geolocation prerequisites to the certified/18.9 branch, the commit that moved res_pjsip documentation to separate XML files was cherry-picked after other that assumed it was already there. This caused a few configuration options to be dropped from the XML and prevented res_stir_shaken and some pjsip transports to fail to load. Those entries are now in their correct spots in pjsip_config.xml. A housekeeping change was also made to both pjsip_config.xml and pjsip_manager.xml to add stylesheet and namespace info. Change-Id: I0ea2eb191e006673f4fa3caafb2edbc92b3e1214	3 years ago
George Joseph	b17e95a792	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
George Joseph	39b9607d4d	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	3 years ago
Kevin Harwell	2074cf07f6	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
Ben Ford	46f5a9c1ae	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
Alexander Traud	f946b92553	progdocs: Fix for Doxygen, the hidden parts. ... ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b	4 years ago
Alexander Traud	b290bb1251	stir/shaken: Avoid a compiler extension of GCC. ... ASTERISK-29776 Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73	4 years ago
Alexander Traud	bae495601a	res_pjsip: Fix for Doxygen. ... ASTERISK-29747 Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084	4 years ago
Josh Soref	dcf492e7b6	res: Spelling fixes ... Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Ben Ford	2e55c0fded	STIR/SHAKEN: Option split and response codes. ... The stir_shaken configuration option now has 4 different choices to pick from: off, attest, verify, and on. Off and on behave the same way they do now. Attest will only perform attestation on the endpoint, and verify will only perform verification on the endpoint. Certain responses are required to be sent based on certain conditions for STIR/SHAKEN. For example, if we get a Date header that is outside of the time range that is considered valid, a 403 Stale Date response should be sent. This and several other responses have been added. Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7	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
Bernd Zobl	804788037e	res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter ... Set preferred transport when querying the local address to use in filter_on_tx_messages(). This prevents the module to erroneously select the wrong transport if more than one transports of the same type (TCP or TLS) are configured. ASTERISK-29241 Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6	4 years ago
Joshua C. Colp	16e4a9d8cf	res_pjsip: On partial transport reload also move factories. ... For connection oriented transports PJSIP uses factories to produce transports. When doing a partial transport reload we need to also move the factory of the transport over so that anything referencing the transport (such as an endpoint) has the factory available. ASTERISK-29441 Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161	4 years ago
George Joseph	655ee680cd	res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs ... RFC7616 and RFC8760 allow more than one WWW-Authenticate or Proxy-Authenticate header per realm, each with different digest algorithms (including new ones like SHA-256 and SHA-512-256). Thankfully however a UAS can NOT send back multiple Authenticate headers for the same realm with the same digest algorithm. The UAS is also supposed to send the headers in order of preference with the first one being the most preferred. We're supposed to send an Authorization header for the first one we encounter for a realm that we can support. The UAS can also send multiple realms, especially when it's a proxy that has forked the request in which case the proxy will aggregate all of the Authenticate headers and then send them all back to the UAC. It doesn't stop there though... Each realm can require a different username from the others. There's also nothing preventing each digest algorithm from having a unique password although I'm not sure if that adds any benefit. So now... For each Authenticate header we encounter, we have to determine if we support the digest algorithm and, if not, just skip the header. We then have to find an auth object that matches the realm AND the digest algorithm or find a wildcard object that matches the digest algorithm. If we find one, we add it to the results vector and read the next Authenticate header. If the next header is for the same realm AND we already added an auth object for that realm, we skip the header. Otherwise we repeat the process for the next header. In the end, we'll have accumulated a list of credentials we can pass to pjproject that it can use to add Authentication headers to a request. NOTE: Neither we nor pjproject can currently handle digest algorithms other than MD5. We don't even have a place for it in the ast_sip_auth object. For this reason, we just skip processing any Authenticate header that's not MD5. When we support the others, we'll move the check into the loop that searches the objects. Changes: * Added a new API ast_sip_retrieve_auths_vector() that takes in a vector of auth ids (usually supplied on a call to ast_sip_create_request_with_auth()) and populates another vector with the actual objects. * Refactored res_pjsip_outbound_authenticator_digest to handle multiple Authenticate headers and set the stage for handling additional digest algorithms. * Added a pjproject patch that allows them to ignore digest algorithms they don't support. This patch has already been merged upstream. * Updated documentation for auth objects in the XML and in pjsip.conf.sample. * Although res_pjsip_authenticator_digest isn't affected by this change, some debugging and a testsuite AMI event was added to facilitate testing. Discovered during OpenSIPit 2021. ASTERISK-29397 Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281	5 years ago
Sean Bright	d2dcd15bd8	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	5 years ago
Joshua C. Colp	65a4a3a4e6	res_pjsip: Give error when TLS transport configured but not supported. ... Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	5 years ago
Joshua C. Colp	f213833514	res_pjsip: Add support for partial transport reload. ... Some configuration items for a transport do not result in the underlying transport changing, but instead are just state we keep ourselves and use. It is perfectly reasonable to change these items. These include local_net and external_* information. ASTERISK-29354 Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a	5 years ago
Alexander Traud	a25bcf70ed	pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. ... Otherwise, Clang 10 warned because of logical-not-parentheses. Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79	5 years ago
Nick French	3c8598ffef	res_pjsip: Prevent segfault in UDP registration with flow transports ... Segfault occurs during outbound UDP registration when all transport states are being iterated over. The transport object in the transport is accessed, but flow transports have a NULL transport object. Modify to not iterate over any flow transport ASTERISK-29210 #close Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a	5 years ago
Alexander Greiner-Baer	c79bd583d9	res_pjsip: set Accept-Encoding to identity in OPTIONS response ... RFC 3261 says that the Accept-Encoding header should be present in an options response. Permitted values according to RFC 2616 are only compression algorithms like gzip or the default identity encoding. Therefore "text/plain" is not a correct value here. As long as the header is hard coded, it should be set to "identity". Without this fix an Alcatel OmniPCX periodically logs warnings like "[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed" on a SIP Trunk. ASTERISK-29165 #close Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840	5 years ago
George Joseph	569fc28966	pjsip_scheduler.c: Add type ONESHOT and enhance cli show command ... * Added a ONESHOT type that never reschedules. * Added "like" capability to "pjsip show scheduled_tasks" so you can do the following: CLI> pjsip show scheduled_tasks like outreg PJSIP Scheduled Tasks: Task Name Interval Times Run ... ============================================= ========= ========= ... pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ... pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ... * Fixed incorrect display of "Next Start". * Compacted the displays of times in the CLI. * Added two new functions (ast_sip_sched_task_get_times2, ast_sip_sched_task_get_times_by_name2) that retrieve the interval, next start time, and next run time in addition to the times already returned by ast_sip_sched_task_get_times(). Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3	5 years ago
Alexander Traud	5a6037778b	res_pjsip/config_transport: Load and run without OpenSSL. ... ASTERISK-28933 Reported-by: Walter Doekes Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f	5 years ago
Kevin Harwell	439f7bb848	res_pjsip, res_pjsip_session: initialize local variables ... This patch initializes a couple of local variables to some default values. Interestingly, in the 'pj_status_t dlg_status' case the value not being initialized caused memory to grow, and not be recovered, in the off nominal path (at least on my machine). Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7	5 years ago
Joshua C. Colp	412b385de5	res_pjsip: Adjust outgoing offer call pref. ... This changes the outgoing offer call preference default option to match the behavior of previous versions of Asterisk. The additional advanced codec negotiation options have also been removed from the sample configuration and marked as reserved for future functionality in XML documentation. The codec preference options have also been fixed to enforce local codec configuration. ASTERISK-29109 Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2	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
Joshua C. Colp	a3d87f78ed	res_pjsip: Fix codec preference defaults. ... When reading in a codec preference configuration option the value would be set on the respective option before applying any default adjustments, resulting in the configuration not being as expected. This was exposed by the REST API push configuration as it used the configuration returned by Asterisk to then do a modification. In the case of codec preferences one of the options had a transcode value of "unspecified" when the defaults should have ensured it would be "allow" instead. This also renames the options in other places that were missed. Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964	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
Joshua C. Colp	4f86118bd8	res_pjsip: Apply AOR outbound proxy to static contacts. ... The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	5 years ago
Roger James	c8dec423d2	pjsip_resolver.c: Ensure AAAA dns requests are made. ... 1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups when an IPV6 transport type has been requested. 2. Rename all occurrences of pjsip_transport_get_type_name to pjsip_transport_get_type_desc. This ensures that the log/debug info shows whether the transport is IPv6 or IPv4. 3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport types. This results in invalid values. Use a bitwise or instead. ASTERISK-26780 Patches: pjsip_resolver.c uploaded by Peter Sokolov (License #7070) Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58	6 years ago
Joshua C. Colp	d845464c76	res_pjsip: Don't set endpoint to unavailable in all cases. ... When an AOR is modified endpoints are updated that reference the AOR so they can start receiving updates and reflect the correct state. If this is the case then we shouldn't change the endpoint to be offline if it does not reference the AOR but instead only when the endpoint is completely updated for all its AORs. ASTERISK-28056 patches: pjsip_options-aor.diff submitted by jhord (license 6978) Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164	6 years ago
George Joseph	7ba6d43083	test_res_pjsip_session_caps: Create unit test ... This unit test runs through combinations of... * Local codecs * Remote Codecs * Codec Preference * Incoming/Outgoing A few new APIs were created to make it easier to test the functionality but didn't result in any actual functional change. ASTERISK_28777 Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164	6 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option ... Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Joshua C. Colp	d6712790cd	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	6 years ago
George Joseph	cbc1136704	res_pjsip_nat: Restore original contact for REGISTER responses ... RFC3261 Section 10 "Registrations", specifically paragraph "10.2.4: Refreshing Bindings", states that a user agent compares each contact address (in a 200 REGISTER response) to see if it created the contact. If the Asterisk endpoint has the rewrite_contact option set however, the contact host and port sent back in the 200 response will be the rewritten one and not the one sent by the user agent. This prevents the user agent from matching its own contact. Some user agents get very upset when this happens and will not consider the registration successful. While this is rare, it is acceptable behavior especially if more than 1 user agent is allowed to register to a single endpoint/aor. This commit updates res_pjsip_nat (where rewrite_contact is implemented) to store the original incoming Contact header in a new "x-ast-orig-host" URI parameter before rewriting it, and to restore the original host and port to the Contact headers in the outgoing response. This is only done if the request is a REGISTER and rewrite_contact is enabled. pjsip_message_filter was also updated to ensure that if a request comes in with any existing x-ast-* URI parameters, we remove them so they don't conflict. Asterisk will never send a request with those headers in it but someone might just decide to add them to a request they craft and send to Asterisk. NOTE: If a device changes its contact address and registers again, it's a NEW registration. If the device didn't unregister the original registration then all existing behavior based on aor/remove_existing and aor/max_contacts apply. ASTERISK-28502 Reported-by: Ross Beer Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e	6 years ago
Kevin Harwell	bdd785d31c	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
Sean Bright	32ce6e9a06	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
Joshua Colp	86452c9fa4	res_pjsip: Fix multiple of the same contact in "pjsip show contacts". ... The code for gathering contacts could result in the same contact being retrieved and added to the list multiple times. The container which stores the contacts to display will now only allow a contact to be added to it once instead of multiple times. ASTERISK-28228 Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df	6 years ago
Ben Ford	86836e0442	pjsip_options.c: Allow immediate qualifies for new contacts. ... When multiple endpoints try to register close together using the same AOR with qualify_frequency set, one contact would qualify immediately while the other contacts would have to wait out the duration of the timer before being able to qualify. Changing the conditional to check the contact container count for a non-zero value allows all contacts to qualify immediately. Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415	7 years ago
Joshua Colp	9c070a4ae3	Merge "res_pjsip: Added a norefersub configuration setting"	7 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	7 years ago
Alexei Gradinari	fe58bc7bdf	res_pjsip: Fix transport_states ref leak ... Add missing ao2_ref(transport_state, -1) while iterate on a transport_states container. Change-Id: I40e35b5a339121300c80075c30db47201a6c374e	7 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	7 years ago
Joshua Colp	c20b84e35b	Merge "res_pjsip_registrar: blocked threads on reliable transport shutdown take 3"	7 years ago
Sean Bright	101272d0dc	Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses." ... This reverts commit d524ad523d. Reason for revert: This causes Contact and Via headers to have the wrong transport address. ASTERISK-28309 #close Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8	7 years ago