asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	870c724f90	Merge "Dialplan functions: Fix some channel autoservice misuse." into 15	7 years ago
Jenkins2	8a9805fb11	Merge "menuselect/menuselect_curses: Resolves sprintf usage error" into 15	7 years ago
Alexander Traud	f00f097eeb	BuildSystem: Enable autotools in Solaris 11. Because this was the last operating system which required a special case, a version appended to the autotools, the whole version stuff is removed by this change. This simplifies the script ./bootstrap.sh. Hopefully, this gives even broader platform compatibility. ASTERISK-27929 ASTERISK-27926 Change-Id: Id4cf433a1a7fa861d0210e1a2e16ca592b49fd5a	7 years ago
Jenkins2	a2da918b4d	Merge "tcptls.h: Remove redundant SSL_CTX typedef." into 15	7 years ago
Richard Mudgett	4d598931ba	channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. * Removed an unnecessary call to ast_channel_blocker_set() in __ast_read(). ASTERISK-27625 Change-Id: I342168b999984666fb869cd519fe779583a73834	7 years ago
Richard Mudgett	f82d0a9380	channel.c: Fix usage of CHECK_BLOCKING() The CHECK_BLOCKING() macro is used to indicate if a channel's handling thread is about to do a blocking operation (poll, read, or write) of media. A few operations such as ast_queue_frame(), soft hangup, and masquerades use the indication to wake up the blocked thread to reevaluate what is going on. ASTERISK-27625 Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d	7 years ago
Richard Mudgett	da2f32bd85	autoservice: Don't start channel autoservice if the thread is a user interface. Executing dialplan functions from either AMI or ARI by getting a variable could place the channel into autoservice. However, these user interface threads do not handle the channel's media so we wind up with two threads attempting to handle the media. 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: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49	7 years ago
Richard Mudgett	437fe5ac76	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
Richard Mudgett	cf56ec739d	AMI PlayDTMF Action: 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: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	7 years ago
Richard Mudgett	84fa1b9c36	Dialplan functions: Fix some channel autoservice misuse. * Fix off nominal paths leaving the channel in autoservice. * Remove unnecessary start/stop channel autoservice. * Fix channel locking around a channel datastore search. Change-Id: I7ff2e42388064fe3149034ecae57604040b8b540	7 years ago
Richard Mudgett	ce2aad08ee	Fix some doxygen and curly placement. Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8	7 years ago
Matthew Fredrickson	c53fcc7650	menuselect/menuselect_curses: Resolves sprintf usage error Acccording to the man page for sprintf, using the same buffer for output as one used as an input yields undefined behavior. This patch should work around this problem. ASTERISK-27903 Reported-by: Alexander Traud Change-Id: I2213dcb454aff26457e2e4cc9c6821276463ae3a	7 years ago
Richard Mudgett	f0ebfa45b5	tcptls.h: Remove redundant SSL_CTX typedef. It is invalid to typedef something more than once. Though not all gcc compilers on different OS's complain about it. Change-Id: I5a7d4565990c985822d61ce75bde0b45f9870540	7 years ago
Richard Mudgett	917cb28a35	channel: Fix some more unprotected channel flag setting. Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c	7 years ago
Joshua Colp	c14ff9a4a6	Merge "app_confbridge: Enable sending events to participants" into 15	7 years ago
Sam Wierema	b51c4a0438	app_mp3: remove 10 seconds of silence after mp3 playback This patch changes the way asterisk polls output from mpg123, instead of waiting for 10 seconds(when playing an http url) it now uses a timeout of one second and iterates 10 times using this same timeout. The main difference is that for every timeout asterisk receives it now checks if mpg123 is still running before poll again. ASTERISK-27752 Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620	7 years ago
Jenkins2	96d8a3de18	Merge "tests/test_utils: Repair ./configure --with-ssl=PATH." into 15	7 years ago
Jenkins2	901f26c3ef	Merge "res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP." into 15	7 years ago
Joshua Colp	8183184e6f	Merge "chan_iax2: better handling for timeout and EINTR" into 15	7 years ago
Alexander Traud	5cc428f840	tests/test_utils: Repair ./configure --with-ssl=PATH. ASTERISK-27914 Change-Id: Ibcab8f556ee77776f203cff8b06d776a673b7bc4	7 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	7 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	7 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	7 years ago
Jenkins2	c594c7a7ca	Merge "rtp: Don't negotiate dynamic codecs using payload." into 15	7 years ago
Joshua Colp	b300deb20f	Merge "chan_pjsip: Register for "BEFORE_MEDIA" responses" into 15	7 years ago
Jenkins2	2ab19d204c	Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated." into 15	7 years ago
Jenkins2	e22efde64e	Merge "crypto.h: Repair ./configure --with-ssl=PATH." into 15	7 years ago
Joshua Colp	ecc709a89d	Merge "res_crypto: Allow OpenSSL configured with no-deprecated." into 15	7 years ago
Joshua Colp	a29e770305	Merge "res_srtp: Repair ./configure --with-ssl=PATH." into 15	7 years ago
George Joseph	7b1c7f7130	Merge "func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql" into 15	7 years ago
Kevin Harwell	01b1785dee	Merge "AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses." into 15	7 years ago
Kevin Harwell	1354686781	Merge "AST-2018-007: iostreams potential DoS when client connection closed prematurely" into 15	7 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	7 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	7 years ago
Alexander Traud	40074487d8	crypto.h: Repair ./configure --with-ssl=PATH. ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	7 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	7 years ago
Alexander Traud	218ac8e9b2	res_srtp: Repair ./configure --with-ssl=PATH. ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2	7 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	7 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	7 years ago
Alexander Traud	ce8c34c3d1	ooh323c: GCC 8.1 warned about output truncated before terminating nul. ASTERISK-27901 Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7	7 years ago
George Joseph	0022cbbb04	Merge "pjsip_options: handle modification of qualify options in realtime" into 15	7 years ago
George Joseph	2a7a439476	Merge "pjsip_options: show/reload AOR qualify options using CLI" into 15	7 years ago
George Joseph	df4d090967	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response" into 15	7 years ago
Joshua Colp	8a925e45a2	Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway" into 15	7 years ago
Joshua Colp	f0b4cd4ed8	Merge "tcptls: Allow OpenSSL configured with no-dh." into 15	7 years ago
George Joseph	302a04571c	Merge "tcptls.h: Repair ./configure --with-ssl=PATH." into 15	7 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	7 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	7 years ago
George Joseph	1d7ade9bbc	Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated." into 15	7 years ago
Joshua Colp	46bff42d88	Merge "app_meetme: Fix manager event documentation for several events." into 15	7 years ago

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