asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	aeff1f2c53	res_musiconhold: Avoid spurious warning when 'format' is the empty string The change to res_config_odbc that allowed empty strings to be returned to realtime consumers¹ causes a warning to be emitted when loading MoH classes. So we need to treat an empty 'format' as if it was not specified to avoid the warning. ASTERISK-28735 #close Reported by: Ross Beer [1] https://gerrit.asterisk.org/c/asterisk/+/13722 Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5	6 years ago
Joshua Colp	49809cb078	Merge "res_rtp_asterisk: Don't produce transport-cc if no packets."	6 years ago
Friendly Automation	dda7f986d0	Merge "res_config_odbc: Preserve empty strings returned by the database"	6 years ago
Friendly Automation	a41c971371	Merge "res_stasis_playback: Prevent media_index from going out of bounds"	6 years ago
Joshua C. Colp	1b53d329ac	res_rtp_asterisk: Don't produce transport-cc if no packets. The code assumed that when the transport-cc feedback function was called at least one packet will have been received. In practice this isn't always true, so now we just reschedule the sending and do nothing. Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a	6 years ago
Joshua Colp	0c845063b3	Merge "res_pjsip_messaging: Allow Content-Type to be overridden"	6 years ago
Friendly Automation	d4b9529a58	Merge "res_stasis: trigger cleanup after update"	6 years ago
Friendly Automation	87ab83d03f	Merge "stasis/app: don't lock an app before a call to send"	6 years ago
Friendly Automation	499f41051e	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating."	6 years ago
Sean Bright	eb9252ea27	res_config_odbc: Preserve empty strings returned by the database When res_config_odbc (and perhaps other realtime backends) reads a SQL NULL from the database, it coalesces the value to the empty string which prevents it from being returned to the realtime core. However, if it instead reads the empty string from the database, it needs a way to encode that fact without having the value omitted entirely. It does this by changing the value to a string with a single space. The realtime code in main/config.c recognizes this special case and _turns the string back into the empty string_ before passing it to realtime API consumers. For all of this to work, we need to ensure that we actually pass the single-space-string back to the realtime core, which is currently failing because we are trimming the value before checking its content. So instead we now special case the single-space-string case so that empty values are returned properly. ASTERISK-28719 #close Reported by: EDV O-TON Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de	6 years ago
Sean Bright	31dc904380	res_stasis_playback: Prevent media_index from going out of bounds Incrementing stasis_app_playback.media_index directly in our playback loop means that when we reach the end of our playlist the index into the vector will be outside of the bounds of the vector. Instead use a temporary variable and only assign when we're sure that we are in bounds. ASTERISK-28713 #close Reported by: Sébastien Duthil Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928	6 years ago
Joshua C. Colp	a1f0c833ab	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	6 years ago
Sean Bright	03d24ca4c1	res_pjsip_messaging: Allow Content-Type to be overridden ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	6 years ago
Kevin Harwell	cce2b0da95	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	6 years ago
Kevin Harwell	4206830a52	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	6 years ago
Sean Bright	b1ca2c5d71	res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly We need to wait for the message sending callback to finish to know if we succeeded or failed. ASTERISK-25421 #close Reported by: Dmitriy Serov Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059	6 years ago
Sean Bright	c376e9f8a8	res_statsd: Document that res_statsd does nothing on its own ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	6 years ago
George Joseph	0380288f7c	Merge "res_realtime: Fix 'realtime update2' argument handling"	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Joshua Colp	d5fce4bc34	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY"	6 years ago
Sean Bright	094e87b0dc	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	6 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	6 years ago
Sean Bright	29d867ed67	res_pjsip_endpoint_identifier_ip: Document support for hostnames ASTERISK-25429 #close Reported by: Joshua C. Colp Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824	6 years ago
Sean Bright	90af050fa4	res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY ASTERISK-27775 #close Reported by: AvayaXAsterisk Change-Id: Iad158e908e34675ad98f74d09c5e73024e50c257	6 years ago
Friendly Automation	51f811183a	Merge "ARI: Ability to inhibit COLP frames when adding channels to a bridge"	6 years ago
Friendly Automation	34746220a0	Merge "res_pjsip_pubsub: Add ability to persist generator state information."	6 years ago
Joshua C. Colp	4e7adbd8f4	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	6 years ago
Sean Bright	312abaa1fe	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	6 years ago
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
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	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

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4984 Commits (4d671a2d14b9b47891075f21a33d3a9a9e5f69ee)