asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	da54605b8a	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	7 years ago
Jenkins2	986cffa18e	Merge "app_confbridge: Enable sending events to participants"	7 years ago
George Joseph	e7a7506f9c	app_confbridge: Enable sending events to participants ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e	7 years ago
Alexander Traud	b01fc2ef3d	res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP. Previously, Asterisk used its script ./configure, to test whether OpenSSL was built with no-srtp (or was simply too old). However, the header file <openssl/opensslconf.h> is the preferred way to detect the local configuration of OpenSSL. As a positive side-effect the script ./configure does not interleave the detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of individual features of OpenSSL anymore. Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0	7 years ago
Jenkins2	8b4c72caea	Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated."	7 years ago
Joshua Colp	782064852a	Merge "res_crypto: Allow OpenSSL configured with no-deprecated."	7 years ago
Jenkins2	a46e658bc0	Merge "res_srtp: Repair ./configure --with-ssl=PATH."	7 years ago
Richard Mudgett	81ac32a85f	AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses. When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. ASTERISK-27818 Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32	7 years ago
Alexander Traud	0743ad6422	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	7 years ago
Alexander Traud	ca682f0030	res_crypto: Allow OpenSSL configured with no-deprecated. The header <openssl/rsa.h> had to be included explicitly. ASTERISK-27906 Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2	7 years ago
Alexander Traud	234bf4b7ff	res_srtp: Repair ./configure --with-ssl=PATH. ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2	7 years ago
George Joseph	7b5fc5d20f	Merge "pjsip_options: handle modification of qualify options in realtime"	7 years ago
Alexei Gradinari	7af5e86821	pjsip_options: show/reload AOR qualify options using CLI Currentrly pjsip_options code does not handle the situation when the AOR qualify options were changed. Also there is no way to find out what qualify options are using. This patch add CLI commands to show and synchronize Aor qualify options: pjsip show qualify endpoint <id> Show the current qualify options for all Aors on the PJSIP endpoint. pjsip show qualify aor <id> Show the PJSIP Aor current qualify options. pjsip reload qualify endpoint <id> Synchronize the qualify options for all Aors on the PJSIP endpoint. pjsip reload qualify aor <id> Synchronize the PJSIP Aor qualify options. ASTERISK-27872 Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c	7 years ago
Alexei Gradinari	e46b442e38	pjsip_options: handle modification of qualify options in realtime Currentrly pjsip_options code does not handle the situation when the qualify options were changed in realtime database. Only 'module reload res_pjsip' helps. This patch add a check on contact add/update observers if the contact qualify options are different than local aor qualify options. If the qualify options were modified then synchronize the pjsip_options AOR local state. ASTERISK-27872 Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62	7 years ago
Joshua Colp	4ea98e49f1	Merge "rtp: Add support for RTP extension negotiation and abs-send-time."	7 years ago
Torrey Searle	c5d2bf05f4	res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change Certain race conditions between changing bridge types and DTMF can cause the current FLAG_NEED_MARKER_BIT to send the marker bit before the actual first packet of native bridging. This logic keeps track of the ssrc the bridge is currently sending and will correctly ensure the marker bit is set if SSRC as changed from the previous sent packet. ASTERISK-27845 Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b	7 years ago
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	7 years ago
Joshua Colp	a507c73a78	rtp: Add support for RTP extension negotiation and abs-send-time. When RTP was originally created it had the ability to place a single extension in an RTP packet. In practice people wanted to potentially put multiple extensions in one and so RFC 5285 (obsoleted by RFC 8285) came into existence. This allows RTP extensions to be negotiated with a unique identifier to be used in the RTP packet, allowing multiple extensions to be present in the packet. This change extends the RTP engine API to add support for this. A user of it can enable extensions and the API provides the ability to retrieve the information (to construct SDP for example) and to provide negotiated information (from SDP). The end result is that the RTP engine can then query to see if the extension has been negotiated and what unique identifier is to be used. It is then up to the RTP engine implementation to construct the packet appropriately. The first extension to use this support is abs-send-time which is defined in the REMB draft[1] and is a second timestamp placed in an RTP packet which is for when the packet has left the sending system. It is used to more accurately determine the available bandwidth. ASTERISK-27831 [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I508deac557867b1e27fc7339be890c8018171588	7 years ago
Matthew Fredrickson	9f9dce05b2	netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04	7 years ago
Alexander Traud	f10fc135d4	res_pjsip_endpoint_identifier_ip: Unregister the module for headers. Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3	7 years ago
Joshua Colp	195af35026	Merge "Fix GCC 8 build issues."	7 years ago
Joshua Colp	a103221de2	Merge "pjsip: Rewrite OPTIONS support with new eyes."	7 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
Matthew Fredrickson	8f55f7c333	res_hep: Adds hostname resolution support for capture_address Previously, only an IP address would be accepted for the capture_address config setting in hep.conf. This change allows capture_address to be a resolvable hostname or an IP address. ASTERISK-27796 #close Reported-By: Sebastian Gutierrez Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f	7 years ago
Jenkins2	57ad488451	Merge "res_ari: Remove requirement that body exists when debug is on."	7 years ago
Jenkins2	8e228fc138	Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."	7 years ago
Jenkins2	0565a6c909	Merge "Add the ability to read the media file type from HTTP header for playback"	7 years ago
Jenkins2	e538fc8e86	Merge "res_rtp_asterisk: Always update SRTP on local SSRC change."	7 years ago
Jenkins2	706b899358	Merge "pjsip: Increase maximum number of usable ciphers & other cleanups"	7 years ago
Joshua Colp	de3ca9bada	res_ari: Remove requirement that body exists when debug is on. The "ari set debug" code for incoming requests incorrectly assumed that all requests would contain a body. If one did not exist the request would be incorrectly rejected. The response that was sent was also incomplete as an incorrect function was used to construct the response. The code has now been changed to no longer require a request to have a body and the response updated to use the correct function. ASTERISK-27801 Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8	7 years ago
Sean Bright	239074c759	pjsip: Increase maximum number of usable ciphers & other cleanups * Increase maximum number of ciphers from 100 to 256 (or whatever PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) * Simplify logic in cipher_name_to_id() * Make signed/unsigned comparison consistent Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 Reported by: Ondřej Holas Change-Id: Iea620f03915a1b873e79743154255c3148a514e7	7 years ago
Richard Mudgett	11b7de82c5	res_pjsip/pjsip_distributor.c: Add missing off-nominal request response. Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b	7 years ago
Richard Mudgett	6cab3c836a	res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests. Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112	7 years ago
Joshua Colp	afdca5c68c	res_rtp_asterisk: Always update SRTP on local SSRC change. When the local SSRC changes we need to update the SRTP information so that the proper key is used. This is commonly done as a result of bridging two channels together. Previously we only updated the SRTP information if media had already flowed, but in practice the channel driver may have already performed SRTP negotiation and set up the previous SSRC. We now always do it on a local SSRC change. ASTERISK-27795 ASTERISK-27800 Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10	7 years ago
Gaurav Khurana	0827d5cc53	Add the ability to read the media file type from HTTP header for playback How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89	7 years ago
George Joseph	3bad41257b	Merge "BuildSystem: Add DragonFly BSD."	7 years ago
Joshua Colp	882e79b77e	pjsip: Rewrite OPTIONS support with new eyes. The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	7 years ago
Jenkins2	9c430569d4	Merge "bridge_softmix: Forward TEXT frames"	7 years ago
Richard Mudgett	661fec4b59	core: Remove unused/incomplete SDP modules. Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86	7 years ago
Jenkins2	56a9338fc1	Merge "Build System: Add missing ASTMM_LIBC to flex output."	7 years ago
Alexander Traud	efe40ff671	BuildSystem: Add DragonFly BSD. ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	7 years ago
Jenkins2	6ccf08c543	Merge "stringfields: Collect extended stringfields into the stringfield section."	7 years ago
Joshua Colp	7b59cfc777	Merge "res_pjsip: Fix deadlock on reliable transport shutdown."	7 years ago
Corey Farrell	179ae87cf4	Build System: Add missing ASTMM_LIBC to flex output. Redirect libc allocation functions to use Asterisk functions for main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors produced by astmm.h when these files are regenerated, though other issues still remain. ASTERISK~27813 Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c	7 years ago
Jenkins2	3255a286b3	Merge "res_musiconhold: Don't restart MOH from beginning after announcement."	7 years ago
George Joseph	4fb7967c73	bridge_softmix: Forward TEXT frames Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	7 years ago
Ben Ford	f5d5083ea7	res_rtp_asterisk: Add support for receiving and handling NACK requests. Adds the ability to receive and handle incoming NACK requests if retransmissions are enabled. If retransmissions are enabled, a data buffer is allocated that stores packets being sent. If a NACK request is received, the packet requested for retransmission is sent if it is still in the buffer. In the same request, if any of the following 16 packets are marked as not received, those will be sent as well if available, as outlined in RFC4585. Also changes RTCP RR and SR to use media source SSRC instead of packet source SSRC when determining which instance to use for RTCP reports. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements ASTERISK-27806 #close Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec	7 years ago
Richard Mudgett	d50d637764	stringfields: Collect extended stringfields into the stringfield section. Use of extended stringfields is a temporary mechanism to avoid ABI breakage in released branches without resorting to more inconvienient methods. * Collect existing extended stringfields into the parent stringfield section of the struct. Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b	7 years ago
George Joseph	38dae51b78	Merge "res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations."	7 years ago
Jenkins2	267e007c28	Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks."	7 years ago

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4591 Commits (eb8bbe660eb3867d3fada24740dc8dc35b384c5d)