3559 Commits (5b7b16a87f5f33ddc100375e4289dfe46ef3c40e)

Author	SHA1	Message	Date
Richard Mudgett	5b7b16a87f	res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. ... * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be	10 years ago
Richard Mudgett	c2ae49249c	res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. ... We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0	10 years ago
Richard Mudgett	2ff26e9746	pjsip_distributor.c: Consistently pick a serializer for messages. ... Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407	10 years ago
Richard Mudgett	df2791da8f	pjsip_distributor.c: Ignore messages until fully booted. ... We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264	10 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	10 years ago
Joshua Colp	216f78c0ce	Merge "Make use of GLOB_BRACE and GLOB_NOMAGIC optional"	10 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	10 years ago
Matt Jordan	7f5ca67e5f	res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded ... A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea	10 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	10 years ago
Joshua Colp	33787459c3	Merge "res_odbc: Implement a connection pool."	10 years ago
Joshua Colp	31a5c28339	res_odbc: Implement a connection pool. ... Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff	10 years ago
Alexander Traud	52120204c9	res_srtp: Instead of libSRTP use OpenSSL as random source. ... Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. Therefore, the symbol RAND_bytes is used instead of crypto_get_random. ASTERISK-24436 #close Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96	10 years ago
George Joseph	a2f820e8dc	ari/resource_channels: Add 'formats' to channel create/originate ... If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b	10 years ago
Timo Teräs	797695c5cc	Make use of GLOB_BRACE and GLOB_NOMAGIC optional ... These flags are non-portable GNU extensions. Make their use optional. This fixes complication error on e.g. musl c-library based systems. Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3	10 years ago
Richard Mudgett	aec7916595	pjsip_distributor.c: Use correct rdata info access method (Part 2). ... The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799	10 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	10 years ago
zuul	dffbb2d7c3	Merge "res_pjsip_mwi_body_generator: Re-order the body items"	10 years ago
Joshua Colp	11ea121cc8	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact"	10 years ago
zuul	853350b5ea	Merge "res_pjsip: Add clarifying documentation to PJSIP_HEADER help text"	10 years ago
zuul	6db496c788	Merge "multicast RTP: Add dialing options"	10 years ago
zuul	47c4f52df6	Merge "res_pjsip: chatty verbose messages"	10 years ago
George Joseph	8a6a14590d	res_pjsip_mwi_body_generator: Re-order the body items ... Re-ordered the body items so Message-Account is second. Messages-Waiting: no Message-Account: sip:1571@<IP Removed>:5060 Voice-Message: 0/0 (0/0) ASTERISK-26065 #close Reported-by: Ross Beer Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3	10 years ago
Rusty Newton	b56f611856	res_pjsip: Add clarifying documentation to PJSIP_HEADER help text ... Added notes about when you can read or write headers. Specifically about being able to read on the inbound channel and write on an outbound channel. ASTERISK-26063 #close Reported by: Private Name Tested by: Rusty Newton Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5	10 years ago
Mark Michelson	bb0f4a6310	multicast RTP: Add dialing options ... This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219	10 years ago
Mark Michelson	88d997913f	ARI: Re-implement the ARI dial command, allowing for early bridging. ... ARI dial had been implemented using the Dial API. This made great sense when dialing was 100% separate from bridging. However, if a channel were to be added to a bridge during the dial attempt, there would be a conflict between the dialing thread and the bridging thread. Each would be attempting to read frames from the dialed channel and act on them. The initial attempt to make the two play nice was to have the Dial API suspend the channel in the bridge and stay in charge of the channel until the dial was complete. The problem with this was that it was riddled with potential race conditions. It also was not well-suited for the case where the channel changed which bridge it was in during the dial. This new approach removes the use of the Dial API altogether. Instead, the channel we are dialing is placed into an invisible ARI dialing bridge. The bridge channel thread handles incoming frames from the channel. If the channel is added to a real bridge, it is departed from the invisible bridge and then added to the real bridge. Similarly, if the channel is removed from the real bridge, it is automatically added back to the invisible bridge if the dial attempt is still active. This approach keeps the threading simple by always having the channel being handled by bridge channel threads. ASTERISK-25925 Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb	10 years ago
Alexei Gradinari	31f17abe44	res_pjsip: add "via_addr", "via_port", "call_id" to contact ... As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	10 years ago
Alexei Gradinari	574c9e77eb	res_pjsip: chatty verbose messages ... There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7	10 years ago
Richard Mudgett	7d44d12816	pjsip_distributor.c: Use correct rdata info access method. ... The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2	10 years ago
zuul	a6b16d7029	Merge "res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying."	10 years ago
zuul	d1ab0936ab	Merge "res_pjsip: Only check transaction on transaction state events."	10 years ago
Joshua Colp	070eab6ed2	res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying. ... Recent changes to res_pjsip_outbound_publish have introduced a race condition at shutdown where an outbound publish may be shutdown twice. In this case the first succeeds as a result of the unpublish. In the second invocation since it's been unpublished a task is queued to just destroy the client. This task holds no ref to the publish and as a result the publish may be destroyed before the task is run, causing a crash. This explicit destruction task now holds a reference to the publish to ensure it remains valid. ASTERISK-26053 #close Change-Id: I10789b98add3e50292ee3b33a55a1d9061cec94b	10 years ago
Joshua Colp	cab97fd905	Merge "ARI: Add the ability to download the media associated with a stored recording"	10 years ago
Joshua Colp	85d0272e76	res_pjsip: Only check transaction on transaction state events. ... The send request callback function currently assumes that it will only ever be called on transaction state changes. This is not always true. If our own timer callback occurs we will call the callback with a timer event instead of a transaction state change event. In this case the transaction on the event is invalid and accessing it will result in a crash. ASTERISK-26049 #close Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc	10 years ago
Mark Michelson	1c02b19b79	res_pjsip: Match dialogs on responses better. ... When receiving an incoming response to a dialog-starting INVITE, we were not matching the response to the INVITE dialog. Since we had not recorded the to-tag to the dialog structure, the PJSIP-provided method to find the dialog did not match. Most of the time, this was not a problem, because there is a fall-back that makes the response get routed to the same serializer that the request was sent on. However, in cases where an asynchronous DNS lookup occurs in the PJSIP core, the thread that sends the INVITE is not actually a threadpool serializer thread. This means we are unable to record a serializer to handle the incoming response. Now, imagine what happens when an INVITE is sent on a non-serialized thread, and an error response (such as a 486) arrives. The 486 ends up getting put on some random threadpool thread. Eventually, a hangup task gets queued on the INVITE dialog serializer. Since the 486 is being handled on a different thread, the hangup task can execute at the same time that the 486 is being handled. The hangup task assumes that it is the sole owner of the INVITE session and channel, so it ends up potentially freeing the channel and NULLing the session's channel pointer. The thread handling the 486 can crash as a result. This change has the incoming response match the INVITE transaction, and then get the dialog from that transaction. It's the same method we had been using for matching incoming CANCEL requests. By doing this, we get the INVITE dialog and can ensure that the 486 response ends up being handled by the same thread as the hangup, ensuring that the hangup runs after the 486 has been completely handled. ASTERISK-25941 #close Reported by Javier Riveros Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0	10 years ago
Matt Jordan	e773e3a9bb	ARI: Add the ability to download the media associated with a stored recording ... This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181	10 years ago
Joshua Colp	40cb032009	res_sorcery_astdb: Filter fields to only the registered ones. ... This change introduces the same filtering that is done in res_sorcery_realtime to the res_sorcery_astdb module. This allows persisted sorcery objects that may contain unknown fields to still be read in from the AstDB and used. This is particularly useful when switching between different versions of Asterisk that may have introduced additional fields. ASTERISK-26014 #close Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2	10 years ago
Joshua Colp	ff3cbc0046	Merge "res_pjsip_empty_info: Respond to empty SIP INFO packets"	10 years ago
Joshua Colp	e205eb55a4	Merge "res_pjsip: Endpoint IP Access Controls"	10 years ago
snuffy	9766a12b4c	res_pjsip_empty_info: Respond to empty SIP INFO packets ... Some SBCs require responses to empty SIP INFO packets after establishing call via INVITE, if not responded to they may drop your call after unspecified timeout of X minutes. They are identified by having no Content-Type, check for this and respond with 200 - OK message. ASTERISK-24986 #close Reported-by: Ilya Trikoz, Federico Santulli Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0	10 years ago
Joshua Colp	b57032c364	Merge "res_hep: Provide an option to pick the UUID type"	10 years ago
Joshua Colp	1f36270b21	Merge "res/res_hep_pjsip: Fix reported local IP address when bound to 'any'"	10 years ago
Joshua Colp	d4b77dad1b	res_pjsip_exten_state: Use the extension for publishing to. ... This change uses the newly added multi-user support for outbound publish to publish to the specific user that an extension state change is for. This also extends the res_pjsip_outbound_publish support to include the user specific From and To URI information in the outbound publishing of extension state. Since the URI is used when constructing the body it is important to ensure that the correct local and remote URIs are used. Finally the max string growths for the dialog-info+xml body generator has been increased as through testing it has proven to be too conservative. ASTERISK-25965 Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1	10 years ago
Kevin Harwell	3905997bae	res_pjsip_outbound_publish: Add multi-user support per configuration ... Added a new multi_user option that when specified allows a particular configuration to be used for multiple users. It does this by replacing the user portion of the server uri with a dynamically created one. Two new API calls have been added in order to make use of the new functionality: ast_sip_publish_user_send - Sends an outgoing publish message based on the given user. If state for the user already exists it uses that, otherwise it dynamically creates new outbound publishing state for the user at that time. ast_sip_publish_user_remove - Removes all outbound publish state objects associated with the user. This essentially stops outbound publishing for the user. ASTERISK-25965 #close Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc	10 years ago
Joshua Colp	e2fc83af50	Merge "ARI: Add the ability to play multiple media URIs in a single operation"	10 years ago
Matt Jordan	03d88b5656	ARI: Add the ability to play multiple media URIs in a single operation ... Many ARI applications will want to play multiple media files in a row to a resource. The most common use case is when building long-ish IVR prompts made up of multiple, smaller sound files. Today, that requires building a small state machine, listening for each PlaybackFinished event, and triggering the next sound file to play. While not especially challenging, it is tedious work. Since requiring developers to write tedious code to do normal activities stinks, this patch adds the ability to play back a list of media files to a resource. Each of the 'play' operations on supported resources (channels and bridges) now accepts a comma delineated list of media URIs to play. A single Playback resource is created as a handle to the entire list. The operation of playing a list is identical to playing a single media URI, save that a new event, PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final media URI. When the entire list is finished being played, a PlaybackFinished event is raised. In order to help inform applications where they are in the list playback, the Playback resource now includes a new, optional attribute, 'next_media_uri', that contains the next URI in the list to be played. It's important to note the following: - If an offset is provided to the 'play' operations, it only applies to the first media URI, as it would be weird to skip n seconds forward in every media resource. - Operations that control the position of the media only affect the current media being played. For example, once a media resource in the list completes, a 'reverse' operation on a subsequent media resource will not start a previously completed media resource at the appropiate offset. - This patch does not add any new operations to control the list. Hopefully, user feedback and/or future patches would add that if people want it. ASTERISK-26022 #close Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f	10 years ago
George Joseph	ae81b55361	res_pjsip_outbound_registration: Clean up state when registration is deleted ... Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23	10 years ago
George Joseph	8b5cee4a4f	res_pjsip: Set TCP_NODELAY on TCP transports ... Although it's perfectly legal to place multiple SIP messages in the same packet, it can cause problems because the Linux default is to enable Path MTU Discovery which sets the Don't Fragment bit on the packets. If adding a second message to the packet causes the MTU to be exceeded, and the destination isn't equipped to send a FRAGMENTATION NEEDED response to a large packet, the packet will just be dropped. We can't specifically tell the stack to send only 1 message per packet, but we can turn on TCP_NODELAY when we create the transport. This will at least tell the stack to send packets as soon as possible. ASTERISK-26005 #close Reported-by: Ross Beer Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd	10 years ago
Matt Jordan	d29c17834c	res/res_hep_pjsip: Fix reported local IP address when bound to 'any' ... When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3	10 years ago
Sean Bright	14938184a3	res_ari: Correct Location headers returned by some ARI resources ... The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	10 years ago
Matt Jordan	e06a23681c	res_hep: Provide an option to pick the UUID type ... At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago