4733 Commits (3077ad0c2402136f44462c3030fb688d0006e830)

Author	SHA1	Message	Date
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Torrey Searle	d0554783e2	res/res_pjsip_nat: Fix logic for REINVITES ... The presence of Record-Route in re-invites is optional, thus it is important to make sure the dialog doesn't have a routset before rewriting the contact header. ASTERISK-28129 #close Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc	7 years ago
Joshua Colp	6f3275e54c	Merge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue"	7 years ago
Chris-Savinovich	a3fc97aa13	res_pjsip: Send a 503 response when overload state if reliable transport. ... When Asterisk's taskprocessors get overloaded we need to reduce the work load. res_pjsip currently ignores new SIP requests and relies on SIP retransmissions in the hope that the overload condition will clear soon enough to handle the retransmitted SIP request. This change adds the following code after ast_taskprocessor_alert_get() has returned TRUE: 1- identifies transport type. If non-udp then send a 503 response 2- if transport type is udp/udp6 then ignore, as before. Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836	7 years ago
Kevin Harwell	fdca9cb64f	res_pjsip: formatting error in documentation ... The use of a '|' in the "global/debug" synopsis documentation caused the generated html table on the wiki to add an extra column that included the text after the pipe. This patch replaces the pipe with a comma. ASTERISK-28150 Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719	7 years ago
Alexei Gradinari	5f3f707793	res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue ... The current round-robin method does not take the current taskprocessor load into consideration when distributing requests. Using the least-size method the request goes to the taskprocessor that is servicing the least number of active tasks at the current time. Longer running tasks with the round-robin method can delay processing tasks. * Change the algorithm from round-robin to least-size for picking the PJSIP taskprocessor from the default serializer pool. Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd	7 years ago
George Joseph	7ece4af59b	Merge "res_pjsip: Add XML documentation for "use_callerid_contact""	7 years ago
Joshua Colp	0c9e217c81	res_pjsip: Add XML documentation for "use_callerid_contact" ... ASTERISK-28087 Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4	7 years ago
Alexei Gradinari	eee935983b	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
George Joseph	584e08b81b	Merge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint"	7 years ago
Alexei Gradinari	e407b8af21	res_pjsip_notify: improve realtime performance on CLI completion on the endpoint ... The module 'res_pjsip_notify' inefficiently makes a lot of DB requests on CLI completion on the endpoint. For example if there are 10k endpoints the module makes 10k requests of these 10k records. Even if a part of the endpoint entered the module makes the same 10k requests and then filtered them by itself. This patch gathers endpoints container by prefix and adds all gathered endpoints to completion at once. ASTERISK-28137 #close Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b	7 years ago
Torrey Searle	cac4ccef25	res_pjsip_session: add new flag use_callerid_contact ... Add a new global flag to res_pjsip to allow the callerid to be used as the username in the contact header. This allows chan_pjsip to have the same behavour as chan_sip ASTERISK-28087 #close Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95	7 years ago
Sean Bright	79c2b4fddd	res_parking: Stop setting the deprecated PARKINGSLOT channel variable. ... Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b	7 years ago
Joshua Colp	dbfb75e02d	Merge "res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability"	7 years ago
George Joseph	a99d48d3f3	Merge "astobj2: Eliminate legacy container allocation macros."	7 years ago
Nick French	37b2e68628	res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability ... This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58	7 years ago
Sean Bright	9e8d671658	res_xmpp: Remove deprecated JabberStatus application. ... Change-Id: I1a00ca22d59d6b6d2166aa56f0e9338a33e5ac60	7 years ago
Richard Mudgett	544ef21bfe	Merge "Fix 'statement' typo throughout code."	7 years ago
Corey Farrell	687ab7aeee	astobj2: Eliminate legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
Richard Mudgett	467f7c6724	Fix 'statement' typo throughout code. ... Most were in comments. A couple were in warning messages. Pointed out by Jonathan H on the Asterisk users mailing list. Change-Id: I6286939dff5d0a27a2758140570106f1cb351855	7 years ago
Richard Mudgett	7ab4befc2b	res_rtp_asterisk.c: Add conditional module dependency to res_pjproject ... * The dependency ensures that res_pjproject cannot be manually unloaded before res_rtp_asterisk. * The dependency allows startup loading errors to report that res_rtp_asterisk depends upon res_pjproject. Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377	7 years ago
Alexei Gradinari	aae5bdc22e	res_pjsip: set callerid_tag to empty string ... This patch sets the callerid_tag to empty string by default. If the callerid_tag is set to NULL then the tag does not become part of a connected line update. For example: Alice's tag is "Alice". Bob's tag is empty. Charlie's tag is "Charlie". Alice calls Bob and then does attended transfer to Charlie. When Alice hangs up the CONNECTEDLINE(tag) is "Alice" on the interception routine on the Charlie's channel, but should be empty. Ths patch also fix memory leaks if there are more then one options "callerid", "callerid_tag", "voicemail_extension" and "contact_user" in the pjsip.conf endpoint definition. Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4	7 years ago
Richard Mudgett	682f96cb5c	res_statsd.c: Fix returned reload status. ... The return status when there was no change in statsd.conf was incorrect. This resulted in the wrong status message on the CLI when reloading the module. * Fixed cleanup on initial load if initializing statsd failed. Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481	7 years ago
George Joseph	0056f260dd	Merge "res_smdi.c: Fix module ref counting and inverted test."	7 years ago
George Joseph	9549511731	Merge "res_statsd.c: Made use defaults if the statsd.conf file does not exist."	7 years ago
Richard Mudgett	3601329c5a	res_smdi.c: Fix module ref counting and inverted test. ... I think this module is so screwed up that it doesn't work anymore. Even with these attempts to fix things it still won't gracefully shut down. The module refs will not go to zero to allow unloading the module. * Fix module ref counting dealing with the SMDI interface object. There were several off-nominal paths that unbalanced the module ref count. Also the destructor freed the ao2 object itself which is bad. Made the smdi_read thread not hold its own ref to the SMDI interface object so when all refs go away the destructor will stop the listener thread. * Fixed the smdi_load() return code of 1 concerning the number of listeners. The test was inverted. Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784	7 years ago
Richard Mudgett	305d08f112	res_smdi.c: Made use defaults if the smdi.conf file does not exist. ... This module is an optional dependency of a couple of other modules. If it declines to load, it then forces other modules that can optionally use this module to also decline. * Made use the default configuration if the config file does not exist and simplified some of the logic. Change-Id: Ib93191f1fe28c0dd9ebe3d84c7762b32f83c4eb9	7 years ago
Richard Mudgett	a69a50b6ec	res_statsd.c: Made use defaults if the statsd.conf file does not exist. ... This module is an optional dependency of many modules. If it declines to load it then forces other modules that can optionally use this module to also decline. * Made use default configuration if there is a config error or the config file does not exist. Change-Id: If1068a582ec54ab7fb437265cb5370a97a825737	7 years ago
Joshua Colp	02bb329bb5	Merge "configure.ac: Check for unbound version >= 1.5"	7 years ago
Joshua Colp	2ac2b5dfb6	Merge "res_pjsip: improve realtime performance on CLI 'pjsip show contacts'"	7 years ago
Joshua Colp	59ad2cc669	Merge "res_stasis: Fix stale data in ARI bridges"	7 years ago
Alexei Gradinari	8bb031abc7	res_pjsip: improve realtime performance on CLI 'pjsip show contacts' ... CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. For example if there are 10k aors then asterisk requests these 10k records of aor and then does 10k requests of contact - one request per aor. Even if use 'like <pattern>' the asterisk requests all aor's and contact's records and then filters them by itself. This patch gathers contact's container by - retrieving all dynamic contacts by regex (filtered by reg_server) - retrieving all aors with permanent contacts - finally filters container by regex ASTERISK-28077 #close Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618	7 years ago
Kevin Harwell	70e4f6f203	Merge "res_odbc: fix missing SQL error diagnostic"	7 years ago
George Joseph	9914e3998e	Merge "res_rtp_asterisk.c: Add "seqno" strictrtp option"	7 years ago
George Joseph	e145b460bb	Merge "res_rtp_asterisk: Raise event when RTP port is allocated"	7 years ago
Moritz Fain	f3422312ea	res_stasis: Fix stale data in ARI bridges ... Fixed an issue that resulted in "Allocation failed" each time an ARI request was made to start playing MOH on a bridge. In bridge_moh_create() we were attaching the after bridge callbacks to chan which is the ;1 channel of the unreal channel pair. We should have attached them to the ;2 channel which is pushed into the bridge by ast_unreal_channel_push_to_bridge(). The callbacks are called when the specific channel leaves the bridging system. Since the ;1 channel is never put into a bridge the callbacks never get called. The callbacks then never remove the moh_wrapper from the app_bridges_moh container. As a result we cannot find the channel associated with the wrapper to start MOH because it has hungup. This is the reason causing the reported issue. * Rather than using after bridge callbacks to cleanup, we now have moh_channel_thread() doing the cleanup when the channel hangs up. * Fixed moh_channel_thread() accumulating control frames on the stasis bridge MOH channel until MOH is stopped. Control frames are no longer accumulated while MOH is playing. * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may or may not return a channel ref. As a result ast_ari_bridges_start_moh() wouldn't know it may have a channel ref to release. stasis_app_bridge_moh_channel() will now return a ref with the channel it returns. * Eliminated RAII_VAR in bridge_moh_create(). ASTERISK-26094 #close Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c	7 years ago
Ben Ford	b11a6643cf	res_rtp_asterisk.c: Add "seqno" strictrtp option ... When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	7 years ago
Alexei Gradinari	e6a69ea2cf	res_odbc: fix missing SQL error diagnostic ... On SQL error there is not diagnostic information about this error. There is only WARNING res_odbc.c: SQL Execute error -1! The function ast_odbc_print_errors calls a SQLGetDiagField to get the number of available diagnostic records, but the SQLGetDiagField returns 0. However SQLGetDiagRec could return one diagnostic records in this case. Looking at many example of getting diagnostics error information I found out that the best way it's to use only SQLGetDiagRec while it returns SQL_SUCCESS. Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR to res_config_odbc. ASTERISK-28065 #close Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1	7 years ago
George Joseph	1ba51b00cc	configure.ac: Check for unbound version >= 1.5 ... In order to do this and provide good feedback, a new macro was created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and path setups for the library then compiles, links and runs a supplied code fragment to do the final determination. In this case, the final code fragment compares UNBOUND_VERSION_MAJOR and UNBOUND_VERSION_MINOR to determine if they're greater than or equal to 1.5. Since we require version 1.5, some code in res_resolver_unbound was also simplified. ASTERISK-28045 Reported by: Samuel Galarneau Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72	7 years ago
Joshua Colp	8bb264841a	res_rtp_asterisk: Raise event when RTP port is allocated ... This change raises a testsuite event to provide what port Asterisk has actually allocated for RTP. This ensures that testsuite tests can remove any assumption of ports and instead use the actual port in use. ASTERISK-28070 Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044	7 years ago
Corey Farrell	93777faf36	json: Take advantage of new API's. ... * Use "o*" format specifier for optional fields in ast_json_party_id. * Stop using ast_json_deep_copy on immutable objects, it is now thread safe to just use ast_json_ref. Additional changes to ast_json_pack calls in the vicinity: * Use "O" when an object needs to be bumped. This was previously avoided as it was not thread safe. * Use "o?" and "O?" to replace NULL with ast_json_null(). The "?" is a new feature of ast_json_pack starting with Asterisk 16. Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48	7 years ago
George Joseph	6658b70ba8	Merge "res_remb_modifier: Add module for controlling REMB from CLI."	7 years ago
George Joseph	ffcccd5e2f	Merge "res_rtp_asterisk: Fix crash on ast_rtp_new failure."	7 years ago
Corey Farrell	bdc8159799	res_rtp_asterisk: Fix crash on ast_rtp_new failure. ... ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call the destroy callback. Remove call to ast_free from ast_rtp_new, leave it to rtp_engine.c to initiate the full cleanup. Add error detection for the ssrc_mapping vector initialization. In rtp_allocate_transport set rtp->s = -1 in the failure path where we close that FD to ensure we don't try closing it twice. ASTERISK-27854 #close Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451	7 years ago
Sean Bright	ad4a6bc27a	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	7 years ago
Sean Bright	a801543f79	AST-2018-009: Fix crash processing websocket HTTP Upgrade requests ... The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. ASTERISK-28013 #close Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a	7 years ago
Joshua Colp	b9874da790	res_remb_modifier: Add module for controlling REMB from CLI. ... This adds a module which registers a CLI command that can set the REMB bitrate value for REMB as it enters or exits Asterisk. This allows you to ignore what Asterisk or a client produces and is useful for demonstrations. This does not generate REMB frames, however, but just modifies them as they flow to or from a channel. Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec	7 years ago
Joshua Colp	e6dcb926fa	Merge "res_pjsip_session: Don't add declined stream if one does not exist."	7 years ago
Joshua Colp	ce9a980be6	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Joshua Colp	32a7b9f4b3	res_pjsip_session: Don't add declined stream if one does not exist. ... Given a scenario where a session refresh was done with a removed stream we would always add a removed stream to the outgoing SDP even if one did not already exist. This change makes it so that a removed stream is only placed into the SDP if one already exists. ASTERISK-28047 Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442	7 years ago