asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	f8b0c2c933	res_pjsip_endpoint_identifier_ip.c: Add port matching support Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	6 years ago
Friendly Automation	7c227aa130	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 16	6 years ago
George Joseph	42eeb6bbe0	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings" into 16	6 years ago
Friendly Automation	86585cbb64	Merge "features.c: Make Bridge application tolerate unspecified channel." into 16	6 years ago
Friendly Automation	077bb05528	Merge "app_dial.c: Simplify dialplan using Dial." into 16	6 years ago
Joshua Colp	5352022b2c	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message." into 16	6 years ago
Friendly Automation	ab042335f5	Merge "app_page.c: Simplify dialplan using Page." into 16	6 years ago
George Joseph	a71f32943c	Merge "app_chanspy.c: Reduce log message level from notice to verbose." into 16	6 years ago
George Joseph	fb7efe34a6	Merge "contrib/valgrind: Fix use of frame-level suppression" into 16	6 years ago
Friendly Automation	09fc6e24e3	Merge "websocket: Consider pending SSL data when waiting for socket input" into 16	6 years ago
Richard Mudgett	a7692ce2f4	app_chanisavail.c: Simplify dialplan using ChanIsAvail. Dialplan has to be careful about passing an empty device list or empty positions in the list. As a result, dialplan has to check for these conditions before using ChanIsAvail. Simplify dialplan by making ChanIsAvail handle these conditions gracefully. * Made tolerate empty positions in the device list. * Simplified the code and eliminated some unnecessary indention. ASTERISK-28638 Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3	6 years ago
Richard Mudgett	144b774b85	app_dial.c: Simplify dialplan using Dial. Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Dial. Simplify dialplan by making Dial handle these conditions gracefully. * Made tolerate empty positions in the dialed device list. * Reduced some message log levels from notice to verbose. ASTERISK-28638 Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9	6 years ago
Richard Mudgett	2780be334d	app_page.c: Simplify dialplan using Page. Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	6 years ago
Richard Mudgett	11753d94d8	features.c: Make Bridge application tolerate unspecified channel. The Bridge application was inconsistent if the channel to bridge with is not specified. If no parameters are given then a warning is issued and the current channel is hung up. If options are given but no channel is specified then a warning is issued and the current channel is not hung up. * Made the Bridge application give a verbose message instead of a warning if the channel to bridge with is not specified and made not hang up the current channel. As a result dialplan no longer needs to check if a channel name is passed before calling Bridge and simply needs to check the BRIDGERESULT channel variable instead. This is something you likely want your dialplan to do anyway. * Fixed up L() option warning message. It is up to the caller to determine if the channel is hung up because of the warning. Dial() hangs up the current channel while Bridge() does not. Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73	6 years ago
Richard Mudgett	00e745066c	app_chanspy.c: Reduce log message level from notice to verbose. Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	198f4cbdbf	app_softhangup.c: Reduce unnecessary warning to verbose message. Why log a warning for something your dialplan explicitly asked for? Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7	6 years ago
Sean Bright	992dcdf780	res_pjsip_config_wizard: Fix change detection for wizard settings ast_sorcery_changeset_create() is not commutative and will fail to detect differences between two variable lists depending on what changed, so switch to ast_variable_lists_match(). ASTERISK-28492 #close Reported by: Jean-Denis Girard Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1	6 years ago
Sean Bright	60fd1322d7	res_agi: Improve GET FULL VARIABLE documentation ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	6 years ago
Sean Bright	47ba42f4a0	websocket: Consider pending SSL data when waiting for socket input When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
Snuffy	b9b50774f5	contrib/valgrind: Fix use of frame-level suppression Fix use of frame-level wildcard usage in suppression file. ASTERISK-27243 #close Reported-by: Richard Kenner Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51	6 years ago
Friendly Automation	3818759e9c	Merge "func_odbc: acf_odbc_read() and cli_odbc_read() unicode support" into 16	6 years ago
Friendly Automation	a992180091	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 16	6 years ago
Boris P. Korzun	e54299cd3e	func_odbc: acf_odbc_read() and cli_odbc_read() unicode support Added ast_odbc_ast_str_SQLGetData() considers SQL_DESC_OCTET_LENGTH column attribute for correct allocating the buffer. ASTERISK-28497 #close Change-Id: I50e86c8a277996f13d4a4b9b318ece0d60b279bf	6 years ago
George Joseph	66ae6f7194	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP" into 16	6 years ago
Jean Aunis	82a870c8c7	chan_sip: voice frames are no longer transmitted after emitting a COLP The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Sean Bright	efa13eb0a0	db: Initialize condition primitive before use The db_init() function ultimately calls db_sync() which signals the condition before it is initialized. Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab	6 years ago
George Joseph	c6dc24fc8e	Merge "config.c: Skip UTF-8 BOMs if present when reading config files" into 16	6 years ago
Joshua C. Colp	3e0a2485c4	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR." into 16	6 years ago
Friendly Automation	2770830495	Merge "main/file.c: Limit media cache usage to remote files." into 16	6 years ago
Friendly Automation	7172230cd0	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors." into 16	6 years ago
Friendly Automation	a3edac10a6	Merge "confbridge: Add support for specifying maximum sample rate." into 16	6 years ago
Sean Bright	a78758d0a2	config.c: Skip UTF-8 BOMs if present when reading config files ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
George Joseph	4b658c82b9	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read" into 16	6 years ago
Joshua C. Colp	a55566b680	Merge "sip_to_pjsip.py: Fix trustrpid typo" into 16	6 years ago
Joshua C. Colp	82e2f0c662	Merge "configure: Add check for MySQL client bool and my_bool type usage." into 16	6 years ago
Friendly Automation	d11534515b	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP." into 16	6 years ago
Pascal Cadotte Michaud	b8e635916f	sip_to_pjsip.py: Fix trustrpid typo ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	6 years ago
George Joseph	b058e2b409	Merge "res_pjsip_nat: Restore original contact for REGISTER responses" into 16	6 years ago
Joshua C. Colp	7167fd6d46	configure: Add check for MySQL client bool and my_bool type usage. Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Friendly Automation	298dd7832b	Merge "PJSIP_CONTACT: add missing argument documentation" into 16	6 years ago
Joshua Colp	8af35e7aa4	Merge "ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging." into 16	6 years ago
Joshua C. Colp	186c4e9b36	res_pjsip_session: Set stream state on created streams for incoming SDP. A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Kevin Harwell	d17bbcb9f1	res_fax: wrap v21 detected Asterisk initiated negotiation with config option A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
Frederic LE FOLL	aa06c6ea29	app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. Temporary channel lifespan is very short and CDR deactivation request through ast_cdr_set_property() may happen when CDR is not available yet. Use CDR_PROP() dialplan function instead, it will first wait for pending CDR insertion requests to be processed. ASTERISK-28636 Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074	6 years ago
Asterisk Development Team	9240fcd8bb	Update CHANGES and UPGRADE.txt for 16.7.0	6 years ago
Jaco Kroon	77941efad9	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Pascal Cadotte Michaud	2d2b28bfa4	PJSIP_CONTACT: add missing argument documentation add missing argument "rtt" and "status" to the documentation The change to the dtd file allow an enumlist to contain one or many configOptionToEnum or enum. This is different from the previous patch I submitted when you could have a configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or (one or many enums) ASTERISK-28626 Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d	6 years ago
George Joseph	4631e77078	Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" into 16	6 years ago
Joshua Colp	9c29c3fb3e	Revert "PJSIP_CONTACT: add missing argument documentation" This reverts commit `174e6426aa`. Reason for revert: Regression in XML validation. validity error : Content model of enumlist is not determinist: (configOptionToEnum \| (configOptionToEnum , enum+) \| enum+) As we are preparing to do releases and this is not critical I am reverting this for now until resolved. Change-Id: I2c9656fb40b2d2f56f54caa35c8be02cc92babd0	6 years ago

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