31418 Commits (c2f81cf446c0338536d2a60bfe91c80caea3aa8c)

 

All Branches   

Search

Author	SHA1	Message	Date
Alexander Traud	99aed78078	crypto.h: Repair ./configure --with-ssl=PATH. ... ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	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
Alexei Gradinari	65ff2f057a	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	1725eaf8fb	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	9f2eb17005	ooh323c: GCC 8.1 warned about output truncated before terminating nul. ... ASTERISK-27901 Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7	7 years ago
George Joseph	7b5fc5d20f	Merge "pjsip_options: handle modification of qualify options in realtime"	7 years ago
George Joseph	fdc931ffc6	Merge "pjsip_options: show/reload AOR qualify options using CLI"	7 years ago
George Joseph	49196c3a7b	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response"	7 years ago
George Joseph	13ac76a230	Merge "cdr_mysql: my_connect_db(): reduce indentation"	7 years ago
George Joseph	5978883d13	Merge "cdr_mysql: split mysql init out of my_load_module"	7 years ago
Joshua Colp	c98c1b3f74	Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway"	7 years ago
Joshua Colp	2151903a16	Merge "tcptls: Allow OpenSSL configured with no-dh."	7 years ago
George Joseph	76339b1962	Merge "tcptls.h: Repair ./configure --with-ssl=PATH."	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
George Joseph	99aad2f0af	Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated."	7 years ago
Joshua Colp	8b08a8437e	Merge "app_meetme: Fix manager event documentation for several events."	7 years ago
Pirmin Walthert	e078558038	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	7 years ago
George Joseph	437ab41881	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	7 years ago
William McCall	a7f4121238	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	7 years ago
Joshua Colp	227f59fdac	Merge "ast_coredumper: Fix output directory and variable precedence"	7 years ago
Richard Mudgett	6bbede84fb	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	7 years ago
Joshua Colp	c63cd006ba	Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated."	7 years ago
Alexander Traud	24503fb600	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	7 years ago
Alexander Traud	d36338ce2b	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	91616f4524	tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. ... ASTERISK-27874 Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd	8 years ago
Joshua Colp	4ea98e49f1	Merge "rtp: Add support for RTP extension negotiation and abs-send-time."	8 years ago
Joshua Colp	1a11fd5187	Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change"	8 years ago
George Joseph	2bf26ce5ac	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	fbb33ba6e8	Merge "tcptls: Repair ./configure --with-ssl=PATH."	8 years ago
Joshua Colp	f5c1e74524	Merge "app_queue: Update year Copyright and fix missing tabs in documentation"	8 years ago
Joshua Colp	ca9120a1f0	Merge "config.c: Fix successful DELETE treated as failure"	8 years ago
Joshua Colp	7e655b26d1	Merge "channel.c: Fix off nominal channel allocation failure path."	8 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	8 years ago
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	8 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	8 years ago
Richard Mudgett	1bec0c73b3	channel.c: Fix off nominal channel allocation failure path. ... __ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd descriptors to -1 yet. The destructor would then attempt to close these fd's that had never been opened. Change-Id: Icf21093f36c60781e8cf6ee9d586536302af33e3	8 years ago
Rodrigo Ramírez Norambuena	d402594f74	app_queue: Update year Copyright and fix missing tabs in documentation ... Change-Id: Ieb8faf37dc765463ee5dbca1d1343242c756b1c7	8 years ago
Alexei Gradinari	39632c7e00	config.c: Fix successful DELETE treated as failure ... The config engine destroy_func callback function returns the number of rows deleted or -1 on error. But the function ast_destroy_realtime_fields treated non-zero return values as error. ASTERISK-27863 Change-Id: Ied02b38e8196cb03043e609a0679feebd288d17b	8 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	8 years ago
Joshua Colp	21dd609e77	Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database"	8 years ago
Alexander Traud	1424f42d25	libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. ... Use CRYPTO_set_id_callback(.) only with OpenSSL 0.9.8 and older. ASTERISK-27867 Change-Id: Iadd58d5bf6f538eb224203970a4e88e26f259655	8 years ago
Alexander Traud	2228ae3f27	tcptls: Repair ./configure --with-ssl=PATH. ... SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH. ASTERISK-27865 Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71	8 years ago
Kevin Harwell	be9a1952d8	Merge "chan_mobile: support handling of caller-id names ("cnam")."	8 years ago
Kevin Harwell	01f3483e6b	Merge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail"	8 years ago
Kevin Harwell	357654313f	Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."	8 years ago
Nic Colledge	2ca3b6d9cc	app_voicemail: Fix data-type mismatch between app_voicemail and database ... Fix data-type mismatch between app_voicemail and database columns exposed by new version of MariaDB ASTERISK-27760 Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b	8 years ago
Nic Colledge	97f20fe5ed	app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail ... Correct the log warning message shown when ODBC voicemail retrieve_file is called and there is a null value in the category column. A more meaningfull message is now written at debug level. ASTERISK-27853 Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4	8 years ago
Brian P. Martin	52ed6bcc8f	chan_mobile: support handling of caller-id names ("cnam"). ... Add support to handle caller-ID names ("cnam") in addition to caller-ID numbers. The prior code ignored the caller-ID name altogether, and used the local name for the cell phone (e.g. "my-iphone") in its place. Note: as of this writing, at least some Android phones don't pass cnam to us. This can be seen by issuing "core set debug 2" in the CLI and watching the "CLIP" record when a call comes in. If cnam isn't in the CLIP record, there's nothing we can do to provide one. We'll provide a null cnam field, so later Asterisk processes know to try other sources (e.g. cidname database, OpenCNAM, etc.). Reported by: Brian Martin Tested by: Brian Martin ASTERISK-27726 Change-Id: I89490d85fa406c36261879c50ae5e65595538ba5	8 years ago