asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	901f26c3ef	Merge "res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP." into 15	8 years ago
Joshua Colp	8183184e6f	Merge "chan_iax2: better handling for timeout and EINTR" into 15	8 years ago
Alexander Traud	5cc428f840	tests/test_utils: Repair ./configure --with-ssl=PATH. ASTERISK-27914 Change-Id: Ibcab8f556ee77776f203cff8b06d776a673b7bc4	8 years ago
George Joseph	8d08b115c5	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	8 years ago
Alexander Traud	d7beaafdd9	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	8 years ago
ktyerman	cdeac34dbe	chan_iax2: better handling for timeout and EINTR The iax2 module is not handling timeout and EINTR case properly. Mainly when there is an interupt to the kernel thread. In case of ast_io_wait recieves a signal, or timeout it can be an error or return 0 which eventually escapes the thread loop, so that it cant recieve any data. This then causes the modules receive queue to build up on the kernel and stop any communications via iax in asterisk. The proposed patch is for the iax module, so that timeout and EINTR does not exit the thread. ASTERISK-27705 Reported-by: Kirsty Tyerman Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb	8 years ago
Jenkins2	c594c7a7ca	Merge "rtp: Don't negotiate dynamic codecs using payload." into 15	8 years ago
Joshua Colp	b300deb20f	Merge "chan_pjsip: Register for "BEFORE_MEDIA" responses" into 15	8 years ago
Jenkins2	2ab19d204c	Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated." into 15	8 years ago
Jenkins2	e22efde64e	Merge "crypto.h: Repair ./configure --with-ssl=PATH." into 15	8 years ago
Joshua Colp	ecc709a89d	Merge "res_crypto: Allow OpenSSL configured with no-deprecated." into 15	8 years ago
Joshua Colp	a29e770305	Merge "res_srtp: Repair ./configure --with-ssl=PATH." into 15	8 years ago
George Joseph	7b1c7f7130	Merge "func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql" into 15	8 years ago
Kevin Harwell	01b1785dee	Merge "AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses." into 15	8 years ago
Kevin Harwell	1354686781	Merge "AST-2018-007: iostreams potential DoS when client connection closed prematurely" into 15	8 years ago
Richard Mudgett	f597032e83	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	8 years ago
Alexander Traud	4b25cde9b3	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	8 years ago
Alexander Traud	40074487d8	crypto.h: Repair ./configure --with-ssl=PATH. ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	8 years ago
Alexander Traud	016070fd3a	res_crypto: Allow OpenSSL configured with no-deprecated. The header <openssl/rsa.h> had to be included explicitly. ASTERISK-27906 Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2	8 years ago
Alexander Traud	218ac8e9b2	res_srtp: Repair ./configure --with-ssl=PATH. ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2	8 years ago
Alexei Gradinari	614d5a5645	func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql The functions acf_odbc_read/cli_odbc_read ignore a number of columns returned by the SQLNumResultCols. If the number of columns is zero it means no data. In this case, a SQLFetch function has to be not called, because it will cause an error. ASTERISK-27888 #close Change-Id: Ie0f7bdac6c405aa5bbd38932c7b831f90729ee19	8 years ago
George Joseph	117141ad3d	chan_pjsip: Register for "BEFORE_MEDIA" responses chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant it was not updating HANGUPCAUSE for 4XX responses. If the remote end sent a "180 Ringing", then a "486 Busy", the hangup cause was left at "180 Normal Clearing". * Removed chan_pjsip_incoming_response from the original session supplement (which was handling only "AFTER MEDIA") and added it to a new session supplement which accepts both "BEFORE_MEDIA" and "AFTER_MEDIA". * Also cleaned up some cleanup code in load module. ASTERISK-27902 Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a	8 years ago
Alexander Traud	ce8c34c3d1	ooh323c: GCC 8.1 warned about output truncated before terminating nul. ASTERISK-27901 Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7	8 years ago
George Joseph	0022cbbb04	Merge "pjsip_options: handle modification of qualify options in realtime" into 15	8 years ago
George Joseph	2a7a439476	Merge "pjsip_options: show/reload AOR qualify options using CLI" into 15	8 years ago
George Joseph	df4d090967	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response" into 15	8 years ago
Joshua Colp	8a925e45a2	Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway" into 15	8 years ago
Joshua Colp	f0b4cd4ed8	Merge "tcptls: Allow OpenSSL configured with no-dh." into 15	8 years ago
George Joseph	302a04571c	Merge "tcptls.h: Repair ./configure --with-ssl=PATH." into 15	8 years ago
Alexei Gradinari	5ddca1a4a8	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	8 years ago
Alexei Gradinari	02ea0fde0b	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	8 years ago
George Joseph	1d7ade9bbc	Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated." into 15	8 years ago
Joshua Colp	46bff42d88	Merge "app_meetme: Fix manager event documentation for several events." into 15	8 years ago
Pirmin Walthert	1958c9e158	bridge_channel.c: Fix Deadlock when using Local channels and fax gateway ast_indicate is invoked with the bridge locked. As ast_indicate locks the other end of the bridge as well this can lead to a deadlock in some situations. (Especially when a different thread does the same in the reverse order). This patch calls ast_indicate after unlocking the bridge which fixes the deadlock. Calling ast_indicate with these parameters without locking the bridge should be safe as this is done at different places without a bridge lock. ASTERISK-27094 #close Reported-by: David Brillert Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f	8 years ago
Joshua Colp	4359a386e5	rtp: Don't negotiate dynamic codecs using payload. In Asterisk there are some dynamic codecs that have a fixed payload number. This number was being improperly used to negotiate the codec, instead of using the name and sample rate. This could result in the wrong payload number being negotiated for a codec. This change makes it so that only static payloads will be negotiated using their payload number. ASTERISK-27848 Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570	8 years ago
George Joseph	89fdbc67c9	app_sendtext: Allow content types other than text/plain There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9	8 years ago
William McCall	350da8f21d	app_confbridge: Add talking indicator for ConfBridgeList AMI response When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	8 years ago
Joshua Colp	b1b7be503e	Merge "ast_coredumper: Fix output directory and variable precedence" into 15	8 years ago
Richard Mudgett	88ff525ffc	app_meetme: Fix manager event documentation for several events. The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and MeetmeTalkRequest AMI events were documented with sending out a Usernum header when the User header was actually output. * Change the online documentation to match reality. ASTERISK-27873 ASTERISK-25261 Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39	8 years ago
Joshua Colp	b07b5b9435	Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated." into 15	8 years ago
Alexander Traud	60242e1418	tcptls.h: Repair ./configure --with-ssl=PATH. asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those inclusions got replaced by forward declarations. As side effect, the inclusions got completed. ASTERISK-27878 Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7	8 years ago
Alexander Traud	cef08821e8	tcptls: 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-27876 Change-Id: I62c2aba4a630aefc231b71f646207e8c027d9497	8 years ago
Alexander Traud	5ffdf896d8	tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. ASTERISK-27874 Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd	8 years ago
Joshua Colp	eb52c70f5b	Merge "rtp: Add support for RTP extension negotiation and abs-send-time." into 15	8 years ago
Joshua Colp	32fa1f9379	Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change" into 15	8 years ago
George Joseph	789f042ad8	ast_coredumper: Fix output directory and variable precedence The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set to "/tmp" instead of "/some/directory". Variables set on the command line or that are already in the environment now take predecence over variables set in the config files. ASTERISK-27846 Reported by: Ted G Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387	8 years ago
Joshua Colp	ae8adf088b	Merge "tcptls: Repair ./configure --with-ssl=PATH." into 15	8 years ago
Joshua Colp	d58a8947f4	Merge "config.c: Fix successful DELETE treated as failure" into 15	8 years ago
Torrey Searle	30da5cfba8	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	8 years ago
Joshua Colp	d94dd27667	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	8 years ago

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