4966 Commits (17.2)

Author	SHA1	Message	Date
Kevin Harwell	19b0efce0d	res_stasis: trigger cleanup after update ... The cleanup code in stasis shuts down applications if they are in a deactivated state, and no longer have explicit subscriptions. When registering an app the cleanup code was running before calling 'update'. When it should be executed after 'update' since a call to register may re-activate the app. We don't want it to shutdown before the 'update' otherwise the app won't be re-activated, or registered. This patch makes it so the cleanup code is executed post 'update'. ASTERISK-28679 #close Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b (cherry picked from commit 7031c3b7bd)	7 years ago
Kevin Harwell	78a907e23d	stasis/app: don't lock an app before a call to send ... Calling 'app_send' eventually calls the app's message handler. It's possible for a handler to obtain a lock on another object, and then need/want to lock the app object. If the caller of 'app_send' locks the app object prior to calling then there's a potential for a deadlock, if another thread calls 'app_send' without locking. This patch makes it so 'app_send' is not called with the app object locked in the section of code doing such. ASTERISK-28423 #close Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27 (cherry picked from commit 737bd8365e)	7 years ago
Joshua C. Colp	5b4d900e43	res_pjsip_pubsub: Increment persistence data ref when recreating. ... Each subscription needs to have a reference to the persisted data for it, as well as the main JSON contained within the tree. When recreating a subscription this did not occur and they both shared the same reference. ASTERISK-28714 Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a (cherry picked from commit 35c9332edf)	7 years ago
Sean Bright	deed0f5706	res_statsd: Document that res_statsd does nothing on its own ... ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	7 years ago
Joshua Colp	cf37de8a5d	Merge "res_realtime: Fix 'realtime update2' argument handling" into 17	7 years ago
Friendly Automation	af4e503005	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY" into 17	7 years ago
Sean Bright	3779e76b68	res_realtime: Fix 'realtime update2' argument handling ... The change in 9b99ef50b5 updated the syntax of the 'realtime update2' CLI command but did not correctly update the calls to ast_update2_realtime(). The issue this addresses was originally opened because we aren't allowing a SQL NULL to be set as part of the update, but this is a limitation of the Realtime API and is not a bug. Additionally, this patch: * Corrects the example in the command documentation to reflect 'update2' instead of 'update.' * Fixes the leading spacing of the command documentation. * Checks that the required 'NULL' literal argument is present where we expect it to be. ASTERISK-21794 #close Reported by: Cédric Bassaget Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd	7 years ago
Sean Bright	28275609b3	res_pjsip_endpoint_identifier_ip: Document support for hostnames ... ASTERISK-25429 #close Reported by: Joshua C. Colp Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824	7 years ago
Sean Bright	6898c8e0da	res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY ... ASTERISK-27775 #close Reported by: AvayaXAsterisk Change-Id: Iad158e908e34675ad98f74d09c5e73024e50c257	7 years ago
Kevin Harwell	5dda6d49b0	Merge "res_pjsip_pubsub: Add ability to persist generator state information." into 17	7 years ago
Joshua C. Colp	e17ba921f3	res_pjsip_pubsub: Add ability to persist generator state information. ... Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de	7 years ago
Sean Bright	eac6eb663d	res_pjsip_endpoint_identifier_ip.c: Add port matching support ... Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	7 years ago
George Joseph	09680cdaa4	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings" into 17	7 years ago
George Joseph	3cc8a1bb84	Merge "websocket: Consider pending SSL data when waiting for socket input" into 17	7 years ago
Sean Bright	a0d1197ab3	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	7 years ago
Sean Bright	7038c8676e	res_agi: Improve GET FULL VARIABLE documentation ... ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	7 years ago
Sean Bright	f344356290	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	7 years ago
Friendly Automation	b401891036	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 17	7 years ago
George Joseph	2634e63f4d	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read" into 17	7 years ago
Joshua C. Colp	58a6d4ecf5	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP." into 17	7 years ago
Friendly Automation	c7966269f5	Merge "res_pjsip_nat: Restore original contact for REGISTER responses" into 17	7 years ago
Joshua C. Colp	c9147a759e	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	7 years ago
Kevin Harwell	68724425e4	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	7 years ago
George Joseph	86822722b6	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	7 years ago
Friendly Automation	5fb0d1d562	Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found" into 17	7 years ago
George Joseph	fd14d7302f	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	7 years ago
Friendly Automation	af079f5085	Merge "res_pjsip_outbound_registration: add support for SRV failover" into 17	7 years ago
Friendly Automation	6685a1862c	Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases" into 17	7 years ago
Friendly Automation	478a602e70	Merge "res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled" into 17	7 years ago
Friendly Automation	77b5ec3e7a	Merge "parking: Fall back to parker channel name even if it matches parkee." into 17	7 years ago
Sean Bright	4e057eb9d2	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	7 years ago
Sean Bright	f26e5bacc0	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	7 years ago
Sean Bright	4d56adf8fb	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	7 years ago
Joshua Colp	41d58a4ce2	parking: Fall back to parker channel name even if it matches parkee. ... ASTERISK-28631 Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc	7 years ago
Salah Ahmed	4ac0299bfb	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	7 years ago
Kevin Harwell	8c99930375	res_pjsip_outbound_registration: add support for SRV failover ... ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561	7 years ago
Sean Bright	76ef36fafc	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	7 years ago
Friendly Automation	f1f28aa9e3	Merge "parking: Fix case where we can't get the parker." into 17	7 years ago
George Joseph	16066ce5fc	Merge "various files - fix some alerts raised by lgtm code analysis" into 17	7 years ago
Joshua Colp	0c486e7edf	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	7 years ago
Kevin Harwell	8bc6fa0fbd	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	7 years ago
Joshua Colp	de433cdcaf	parking: Fix case where we can't get the parker. ... ASTERISK-28616 Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d	7 years ago
Joshua Colp	d638d9c6c6	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	7 years ago
Kevin Harwell	3084a6c617	Merge "res_pjsip_session: initialize pending's topology to endpoint's" into 17	7 years ago
Kevin Harwell	ea3daa94c8	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	7 years ago
George Joseph	7202624b3b	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	7 years ago
Joshua Colp	eea2d499f4	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	7 years ago
Joshua Colp	43ea6e21db	Merge "res_calendar: Resolve memory leak on calendar destruction" into 17	7 years ago
Sean Bright	b3792e1288	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	7 years ago
Joshua Colp	e37d546109	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	7 years ago