4955 Commits (a4fd89536d8d5852c6d8337f7bfa23794eddb6ce)

Author	SHA1	Message	Date
George Joseph	d66b01d3bf	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings"	6 years ago
Friendly Automation	255a647c53	Merge "websocket: Consider pending SSL data when waiting for socket input"	6 years ago
Sean Bright	b40dd11afe	res_pjsip_config_wizard: Fix change detection for wizard settings ... ast_sorcery_changeset_create() is not commutative and will fail to detect differences between two variable lists depending on what changed, so switch to ast_variable_lists_match(). ASTERISK-28492 #close Reported by: Jean-Denis Girard Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1	6 years ago
Sean Bright	7d94bdde9d	res_agi: Improve GET FULL VARIABLE documentation ... ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	6 years ago
Sean Bright	87110c1bdf	websocket: Consider pending SSL data when waiting for socket input ... When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
George Joseph	be93537382	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option"	6 years ago
George Joseph	657ada8bfd	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read"	6 years ago
Joshua C. Colp	2779189689	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP."	6 years ago
George Joseph	8587ef3a0c	Merge "res_pjsip_nat: Restore original contact for REGISTER responses"	6 years ago
Joshua C. Colp	a603d7d324	res_pjsip_session: Set stream state on created streams for incoming SDP. ... A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Kevin Harwell	b6f5607359	res_fax: wrap v21 detected Asterisk initiated negotiation with config option ... A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
George Joseph	39c920ac78	res_rtp_asterisk: Add frame list cleanups to ast_rtp_read ... In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a	6 years ago
George Joseph	d8dac784f4	Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found"	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
Friendly Automation	ec559c34db	Merge "res_pjsip_outbound_registration: add support for SRV failover"	6 years ago
Friendly Automation	e99cdeeff3	Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases"	6 years ago
Kevin Harwell	bb2a59e171	Merge "res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled"	6 years ago
Friendly Automation	0f1a429945	Merge "parking: Fall back to parker channel name even if it matches parkee."	6 years ago
Sean Bright	b1be06df8d	res_pjsip_registrar.c: Prevent potential double free if AOR is not found ... The simple fix here is simply to NULL out username and password after we call ast_free on them. Unfortunately, I noticed that we weren't checking for allocation failures for username and password, and adding those checks made things noisy and cumbersome. So instead we partially rollback the recent LGTM patch, and move the alloca calls into find_aor_name(). ASTERISK-28641 #close Reported by: Ross Beer Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2	6 years ago
Sean Bright	0183e2bc67	res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases ... We're appropriately sizing the id_domain_alias buffer, but then copying the data into the id_domain one. We were then using the uninitialized id_domain_alias buffer we just allocated. This is ASTERISK~28641 adjacent, but significant enough to warrant its own patch. Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9	6 years ago
Sean Bright	6ee1f1f507	res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled ... We need to copy the endpoint name before we call ao2_cleanup() on it, otherwise we might try to access memory that has been reclaimed. ASTERISK-28445 #close Reported by: Bernhard Schmidt Change-Id: I404b952608aa606e0babd3c4108346721fb726b3	6 years ago
Joshua Colp	811ae88da4	parking: Fall back to parker channel name even if it matches parkee. ... ASTERISK-28631 Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc	6 years ago
Salah Ahmed	330ffa2bce	res_pjsip_t38: T.38 error correction mode selection at 200 ok received ... if asterisk offer T38 SDP with none error correction scheme and the endpoint respond with redundancy EC scheme, asterisk switch to that mode. Since we configure the endpoint as none EC mode we should not switch to any other mode except none. following logic implemented in code. 1. If asterisk offer none, and anything except none in answer will be ignored. 2. If asterisk offer fec, answer with fec, redundancy and none will be accepted. 3. If asterisk offer redundancy, answer with redundancy and none will be accepted. ASTERISK-28621 Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19	6 years ago
Kevin Harwell	d5d41409e2	res_pjsip_outbound_registration: add support for SRV failover ... ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561	6 years ago
Sean Bright	a5fa0d662e	res_pjsip_registrar: Fix uninitlized variable warning ... Fixes: error: ‘domain_name’ may be used uninitialized in this function Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008 Change-Id: I44413b49ea1205aa25538142161deb73883c79e8	6 years ago
George Joseph	b95bc30c40	Merge "parking: Fix case where we can't get the parker."	6 years ago
George Joseph	b3de3ce042	Merge "res_rtp_asterisk: Always return provided DTLS packet length."	6 years ago
Joshua Colp	02129ad4d0	res_rtp_asterisk: Always return provided DTLS packet length. ... OpenSSL can not tolerate if the packet sent out does not match the length that it provided to the sender. This change lies and says that each time the full packet was sent. If a problem does occur then a retransmission will occur as appropriate. ASTERISK-28576 Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449	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
Joshua Colp	807a70b7ae	parking: Fix case where we can't get the parker. ... ASTERISK-28616 Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d	6 years ago
George Joseph	990a91b44a	stasis: Don't hold app_registry and session locks unnecessarily ... resource_events:stasis_app_message_handler() was locking the session, then attempting to determine if the app had debug enabled which locked the app_registry container. res_stasis:__stasis_app_register was locking the app_registry container then calling app_update which caused app_handler (which locks the session) to run. The result was a deadlock. * Updated resource_events:stasis_app_message_handler() to determine if debug was set (which locks the app_registry) before obtaining the session lock. * Updated res_stasis:__stasis_app_register to release the app_registry container lock before calling app_update (which locks the sesison). ASTERISK-28423 Reported by Ross Beer Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4	6 years ago
Friendly Automation	ad6314c90f	Merge "parking: Use channel snapshot instead of channel."	6 years ago
Joshua Colp	e924c5107c	parking: Use channel snapshot instead of channel. ... There exists a scenario where a thread can hold a lock on the channels container while trying to lock a bridge. At the same time another thread can hold the lock for said bridge while attempting to retrieve a channel. This causes a deadlock. This change fixes this scenario by retrieving a channel snapshot instead of a channel, as information present in the snapshot is all that is needed. ASTERISK-28616 Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2	6 years ago
Kevin Harwell	0e3b397812	res_pjsip_session: initialize pending's topology to endpoint's ... Found during some testing, there is a race condition between selecting an appropriate bridge type for a call versus the applying of media on the callee's session. In some instances a native bridge type would have been chosen, but due to the callee's media not yet being established at bridge compatibility check time the simple bridge type is picked instead. When using chan_pjsip this initiates a topology change event. The topologies are then compared for the two sessions. However, when the topology was created for the caller its streams are initialized to "inactive". This topology is then used as a base when creating the callee's topology, and streams. Soon after the caller's topology's stream(s) get updated based on the sdp (get set to sendrecv in the failing scenario). Now when the topology change event is raised, and the two topologies are compared, the comparison fails due to a stream state mismatch (sendrecv vs inactive). And since they differ a reinvite is sent out (to the caller in this case). This patch makes it such that when the caller's topology is initially created it gets created based on its configured endpoint's media topology. When the endpoint's topology is created its stream's state(s) are initialized to sendrecv instead of inactive. Subsequently, now when the callee's topology is created its topology streams are now initialized to sendrecv. Thus when the topology change event occurs due to the mentioned scenario the stream states match for the given sessions, and the reinvite is not sent unless due to some other valid mismatch. Note, this patch only changes one pending media state's creation point. It's possible other places *could* be changed, however for now it was deemed best to only alter what's here. Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7	6 years ago
Joshua Colp	e73eba85c1	res_pjsip_outbound_registration: Extend documentation for "max_retries". ... If the "max_retries" option is set to 0 then upon failure no further attemps are made, so explicitly document the behavior. ASTERISK-28602 Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf	6 years ago
Joshua Colp	75f6f64058	Merge "res_ari_events: Add module reference when a WebSocket is open."	6 years ago
Sean Bright	16e668c7dd	res_calendar: Resolve memory leak on calendar destruction ... Calling ne_uri_parse allocates memory that needs to be freed with a corresponding call to ne_uri_free. ASTERISK-28572 #close Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61	6 years ago
Joshua Colp	360936ead5	res_ari_events: Add module reference when a WebSocket is open. ... This change ensures that the module isn't unloaded when a WebSocket is open. Previously it was possible to unload the module manually or during shutdown which could cause a crash when any active WebSockets were terminated. ASTERISK-28585 Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c	6 years ago
George Joseph	d5fab6f22e	Merge "ExternalMedia: Change return object from ExternalMedia to Channel"	6 years ago
George Joseph	d71d0f9489	ExternalMedia: Change return object from ExternalMedia to Channel ... When we created the External Media addition to ARI we created an ExternalMedia object to be returned from the channels/externalMedia REST endpoint. This object contained the channel object that was created plus local_address and local_port attributes (which are also in the Channel variables). At the time, we thought that creating an ExternalMedia object would give us more flexibility in the future but as we created the sample speech to text application, we discovered that it doesn't work so well with ARI client libraries that a) don't have the ExternalMedia object defined and/or b) can't promote the embedded channel structure to a first-class Channel object. This change causes the channels/externalMedia REST endpoint to return a Channel object (like channels/create and channels/originate) instead of the ExternalMedia object. Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9	6 years ago
Joshua Colp	6e907ae5d4	res_rtp_asterisk: Remove a log message that slipped in. ... This was only supposed to be for testing, so now it can be removed. Change-Id: I3dfc2e776e70b3196aeed5688372ea80c0214b59	6 years ago
George Joseph	8f67495306	Merge "res_pjsip_mwi: potential double unref, and potential unwanted double link"	6 years ago
Friendly Automation	cb6441446e	Merge "cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12"	6 years ago
Christoph Moench-Tegeder	52ade18420	cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12 ... PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef (column default values), which has been deprecated since version 8.0. Since then, the official/correct/supported way to retrieve the column default value from the catalog is function pg_catalog.pg_get_expr(). This change breaks compatibility with pre-8.0 PostgreSQL servers, but has reached end-of-support more than a decade ago. cdr_pgsql and res_config_pgsql still have support for pre-7.3 servers, but cleaning that up is perhaps a topic for a major release, not this bugfix. ASTERISK-28571 Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe	6 years ago
Kevin Harwell	5dae803eea	res_pjsip_mwi: potential double unref, and potential unwanted double link ... When creating an unsolicited MWI aggregate subscription it was possible for the subscription object to be double unref'ed. This patch removes the explicit unref as it is not needed since the RAII_VAR will handle it at function end. Less concerning there was also a bug that could potentially allow the aggregate subscription object to be added to the unsolicited container twice. This patch ensures it is added only once. ASTERISK-28575 Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763	6 years ago
Kevin Harwell	12dbeb69b0	res_pjsip_mwi: use an ao2_global object for mwi containers ... On shutdown it's possible for the unsolicited mwi container to be freed before other dependent threads are done using it. This patch ensures this can no longer happen by wrapping the container in an ao2_global object. The solicited container was also changed too. ASTERISK-28552 Change-Id: I8f812286dc19a34916acacd71ce2ec26e1042047	6 years ago
Kevin Harwell	2970a13fb8	res_pjsip/res_pjsip_mwi: use centralized serializer pools ... Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they both implemented their own serializer pool functionality that was pretty much identical in each of the source files. This patch removes the duplicated code, and uses the new 'ast_serializer_pool' object instead. Additionally res_pjsip_mwi enables a shutdown group on the pool since if the timing was right the module could be unloaded while taskprocessor threads still needed to execute, thus causing a crash. Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d	6 years ago
Friendly Automation	728a0ebb06	Merge "channel/chan_pjsip: add dialplan function for music on hold"	6 years ago
George Joseph	2e6f1e9602	Merge "res_pjsip_pubsub: add endpoint to some warning"	7 years ago
Torrey Searle	b43cdc7f1e	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	7 years ago