31602 Commits (0db94c7519ff6d12e280c84c26b7f739a1838d0c)

 

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Author	SHA1	Message	Date
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	7 years ago
Alexander Traud	91616f4524	tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. ... ASTERISK-27874 Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd	7 years ago
Joshua Colp	4ea98e49f1	Merge "rtp: Add support for RTP extension negotiation and abs-send-time."	7 years ago
Joshua Colp	1a11fd5187	Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change"	7 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	7 years ago
Joshua Colp	fbb33ba6e8	Merge "tcptls: Repair ./configure --with-ssl=PATH."	7 years ago
Joshua Colp	f5c1e74524	Merge "app_queue: Update year Copyright and fix missing tabs in documentation"	7 years ago
Joshua Colp	ca9120a1f0	Merge "config.c: Fix successful DELETE treated as failure"	7 years ago
Joshua Colp	7e655b26d1	Merge "channel.c: Fix off nominal channel allocation failure path."	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	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
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	8 years ago
Joshua Colp	60ce5d0003	Merge "cli: Display correct unit for HTTP timeout in "manager show settings"."	8 years ago
Joshua Colp	195af35026	Merge "Fix GCC 8 build issues."	8 years ago
Joshua Colp	1cc4313d32	Merge "rtp_engine: Remove the double assigned RTP payload ID of H.263+."	8 years ago
Joshua Colp	c0c79997f8	Merge "git: Ignore *.orig."	8 years ago
Joshua Colp	bc4b3c535d	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code."	8 years ago
Joshua Colp	a103221de2	Merge "pjsip: Rewrite OPTIONS support with new eyes."	8 years ago
Alexander Traud	71d1e8d8c8	rtp_engine: Remove the double assigned RTP payload ID of H.263+. ... Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8 assigned another payload ID 98 for this format in Asterisk 1.6. Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667	8 years ago
Corey Farrell	4722a653f4	cli: Display correct unit for HTTP timeout in "manager show settings". ... HTTP timeout is in seconds, not minutes. ASTERISK-27852 #close Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e	8 years ago
Alexander Traud	263637a38d	rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. ... Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48	8 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	8 years ago
Alexander Traud	919b0eb3f2	rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. ... This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples of media formats which were affected. ASTERISK-27850 Reported by: Dinis Brazão, Selene Feigl Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496	8 years ago
Joshua Colp	6773ea9e39	Merge "makeopts.in: Remove unused/undefined AST_MARCH_NATIVE."	8 years ago
Joshua Colp	5437b3932d	Merge "sip_to_pjsip: Enable python3 compatibility."	8 years ago
Joshua Colp	1351f42363	Merge "res_hep: Adds hostname resolution support for capture_address"	8 years ago
Jenkins2	179c794879	Merge "app_macro: Prevent infinite loop in find_matching_priority."	8 years ago