1183 Commits (bef49d90c1cc013aba248c757cd3db2436175c03)

Author	SHA1	Message	Date
George Joseph	b148453708	Merge "res_pjsip_pubsub: Prune subs with reliable transports at startup"	8 years ago
Corey Farrell	4e4428ef3c	res_pjsip_registrar_expire: Delete empty module. ... Verified nothing in the testsuite lists this module as a dependency. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813	8 years ago
George Joseph	2b9aa6b5bb	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	8 years ago
Richard Mudgett	b9e35bf6d3	CHANGES: Add AMI action 'PJSIPShowContacts' note. ... ASTERISK-27581 Change-Id: If6af275764741a11030f0a4fd324fa29b376d74e	8 years ago
krells	77f2814d01	pbx: Reduce verbosity while loading extensions ... Each time the dial plan is reloaded, a lot of logs like these are generated: "Added extension 'XXXXX' priority 1 to YYYYYYYYYYY" This patch changes the log level for those logs. ASTERISK-27084 Change-Id: I5662902161c50890997ddc56835d4cafb456c529	8 years ago
ghjm	86b484dec7	app_followme: Add a prompt to be read when a call is connected ... This patch adds the ability to configure a prompt which will be read to the "winner" who pressed 1 (or the configured value) and received the call. ASTERISK-24372 #close Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272	8 years ago
Richard Mudgett	8494e78010	res_pjsip: Split type=identify to IP address and SIP header matching priorities ... The type=identify endpoint identification method can match by IP address and by SIP header. However, the SIP header matching has limited usefulness because you cannot specify the SIP header matching priority relative to the IP address matching. All the matching happens at the same priority and the order of evaluating the identify sections is indeterminate. e.g., If you had two type=identify sections where one matches by IP address for endpoint alice and the other matches by SIP header for endpoint bob then you couldn't predict which endpoint is matched when a request comes in that matches both. * Extract the SIP header matching criteria into its own "header" endpoint identification method so the user can specify the relative priority of the SIP header and the IP address matching criteria in the global endpoint_identifier_order option. The "ip" endpoint identification method now only matches by IP address. ASTERISK-27491 Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095	8 years ago
Sean Bright	9e2fcb82ed	cdr_syslog: Deprecate unmaintained module ... There has been an open issue against cdr_syslog (ASTERISK~14441) about a race condition for 7.5 years that has never been addressed. Because this module is effectively unmaintained and currently broken, there is no sense in keeping it around. If logging CDRs to syslog is a desirable feature, it would probably be better to write the logs directly to the syslog server via socket instead of using the facilities provided by openlog/syslog/closelog. Doing so would address the race condition referenced in the associated issue. Change-Id: Ic77b94cd97f355a9cf5b1d3f3444964a6e0ba5dc	8 years ago
Sungtae Kim	faeb9e1b26	res_pjsip: Add AMI action 'PJSIPShowAuths' ... Add an AMI action which provides information on all configured Auths. ASTERISK-27547 Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817	8 years ago
Sungtae Kim	ffbf5be116	res_pjsip: Add AMI action 'PJSIPShowAors' ... Add an AMI action which provides information on all configured AORs. ASTERISK-27537 Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb	8 years ago
Rodrigo Ramírez Norambuena	5335ad117d	app_queue: Add feature to set wrapuptime on the queue member ... This patch adds the ability to set the wrapuptime on the queue member config. When the option is set the wrapuptime on the queue member is used instead of the queue's wrapuptime. ASTERISK-27483 #close Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902	8 years ago
Corey Farrell	6f8b34f9c1	chan_sip: Add security event for calls to invalid extension. ... Log a message to security events when an INVITE is received to an invalid extension. ASTERISK-25869 #close Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f	8 years ago
Kevin Harwell	b088cddc03	pjsip_options: wrongly applied "UNKNOWN" status ... A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89	8 years ago
Richard Mudgett	90bb0a3e10	core: Add cache_media_frames debugging option. ... The media frame cache gets in the way of finding use after free errors of media frames. Tools like valgrind and MALLOC_DEBUG don't know when a frame is released because it gets put into the cache instead of being freed. * Added the "cache_media_frames" option to asterisk.conf. Disabling the option helps track down media frame mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the frame is used after free and who freed it. NOTE: This option has no effect when Asterisk is compiled with the LOW_MEMORY compile time option enabled because the cache code does not exist. To disable the media frame cache simply disable the cache_media_frames option in asterisk.conf and restart Asterisk. Sample asterisk.conf setting: [options] cache_media_frames=no ASTERISK-27413 Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00	8 years ago
Sean Bright	04d3785a79	dtls: Add support for ephemeral DTLS certificates. ... This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4	8 years ago
Igor Goncharovskiy	9bad4c74cc	app_agent_spool: Fix typo in dtmf features usage desctiption ... Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function instead of correct 'dtmf_features' ASTERISK-27377 #close Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930	8 years ago
Joshua Colp	9e1fbab382	res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. ... When the identify_by option on an endpoint is set to ip it will only be identified using the res_pjsip_endpoint_identifier_ip module. This ensures that it is not mistakenly matched using the username of the From header. To ensure behavior has not changed the default has been changed to "username,ip" for the identify_by option. ASTERISK-27206 Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd	8 years ago
Corey Farrell	955a891a84	app_macro deprecation. ... * Mark the module deprecated. * Disable the module by default. * Produce a warning the first time a macro is used. * Note deprecation related options in app_dial and app_queue. ASTERISK-27350 Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc	8 years ago
Thomas Sevestre	21c0283b78	features, manager : Add CancelAtxfer AMI action ... Add action to cancel feature attended transfer with AMI interface ASTERISK-27215 #close Change-Id: Iab8a81362b5a1757e2608f70b014ef863200cb42	8 years ago
Richard Mudgett	fb19799b62	res_pjsip_registrar.c: Update remove_existing AOR contact handling. ... When "rewrite_contact" is enabled, the "max_contacts" count option can block re-registrations because the source port from the endpoint can be random. When the re-registration is blocked, the endpoint may give up re-registering and require manual intervention. * The "remove_existing" option now allows a registration to succeed by displacing any existing contacts that now exceed the "max_contacts" count. Any removed contacts are the next to expire. The behaviour change is beneficial when "rewrite_contact" is enabled and "max_contacts" is greater than one. The removed contact is likely the old contact created by "rewrite_contact" that the device is refreshing. ASTERISK-27192 Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b	8 years ago
George Joseph	446d48fd49	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Andrey Egorov	15fbcc74d8	res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif ... Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db	8 years ago
George Joseph	305bd0d99f	Make --with-pjproject-bundled the default for Asterisk 15 ... '--with-pjproject-bundled' is now the default when running ./configure. It can be disabled with '--without-pjproject-bundled'. To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. ASTERISK-27189 Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce	8 years ago
kkm	4b58609c33	chan_sip: Access incoming REFER headers in dialplan ... This adds a way to access information passed along with SIP headers in a REFER message that initiates a transfer. Headers matching a dialplan variable GET_TRANSFERRER_DATA in the transferrer channel are added to a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH. The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for headers that should be put into the hash. If not set, no headers are included. If set to a string (perhaps 'X-' in a typical case), all headers starting this string are added. Empty string matches all headers. If there are multiple of the same header, only the latest occurrence in the REFER message is available in the hash. Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the referrer channel, and should be set with the '_' or '__' prefix. I avoided a specific reference to SIP or REFER, as in my mind the mechanism can be generalized to other channel techs. ASTERISK-27162 Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e	8 years ago
kkm	4c0798e91d	chan_sip: Add dialplan function SIP_HEADERS ... Syntax: SIP_HEADERS([prefix]) If the argument is specified, only the headers matching the given prefix are returned. The function returns a comma-separated list of SIP header names from an incoming INVITE message. Multiple headers with the same name are included in the list only once. The returned list can be iterated over using the functions POP() and SIP_HEADER(). For example, '${SIP_HEADERS(Co)}' might return the string 'Contact,Content-Length,Content-Type'. Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional extended headers sent by a peer. ASTERISK-27163 Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267	8 years ago
Joshua Colp	2a4283f3e7	res_pjsip: Add support for dnsmgr to external_media_address. ... The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2	8 years ago
Sean Bright	3f98488279	app_queue: Add announce-position-only-up option ... Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6	8 years ago
Torrey Searle	65c560894d	chan_pjsip: add a new function PJSIP_DTMF_MODE ... This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612	8 years ago
George Joseph	b7a875778a	res_musiconhold: Add kill_escalation_delay, kill_method to class ... By default, when res_musiconhold reloads or unloads, it sends a HUP signal to custom applications (and all descendants), waits 100ms, then sends a TERM signal, waits 100ms, then finally sends a KILL signal. An application which is interacting with an external device and/or spawns children of its own may not be able to exit cleanly in the default times, expecially if sent a KILL signal, or if it's children are getting signals directly from res_musiconhoild. * To allow extra time, the 'kill_escalation_delay' class option can be used to set the number of milliseconds res_musiconhold waits before escalating kill signals, with the default being the current 100ms. * To control to whom the signals are sent, the "kill_method" class option can be set to "process_group" (the default, existing behavior), which sends signals to the application and its descendants directly, or "process" which sends signals only to the application itself. Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b	8 years ago
Jenkins2	33aa3907eb	Merge "pjproject_bundled: Allow passing configure options to bundled"	8 years ago
Jenkins2	e1c0e14fac	Merge "res_pjsip: Add DTMF INFO Failback mode"	8 years ago
George Joseph	f573e599c0	pjproject_bundled: Allow passing configure options to bundled ... There wasn't any good way to pass options like --host or --build down to the pjproject configure which makes cross-compiling difficult. * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used to pass arbitrary options to pjproject configure. * Automatically set the pjproject configure --host and --build options to match those supplied for the asterisk configure. ASTERISK-27097 #close Reported-by: Kinsey Moore Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e	8 years ago
Torrey Searle	fb7247c57c	res_pjsip: Add DTMF INFO Failback mode ... The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91	8 years ago
Alexei Gradinari	0cef7b9d4e	app_voicemail: IMAP connection control ... A new global option "imap_poll_logout" was added to specify whether need to disconnect from the IMAP server after polling of mailboxes. ASTERISK-27068 #close Closing IMAP connection after loading mailbox from voicemail.conf ASTERISK-24052 #close Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a	8 years ago
Alexei Gradinari	7a46309d3d	res_pjsip: New endpoint option "notify_early_inuse_ringing" ... This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	8 years ago
Jenkins2	abe1dd3039	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing."	8 years ago
Alexei Gradinari	19da99df2f	CHANGES: correct version for a new option 'refer_blind_progress' ... Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97	9 years ago
Joshua Colp	d3e951edf5	pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. ... PJSIP support in Asterisk differs from chan_sip in that it allows media to be sent as-is without transcoding provided the codecs were negotiated in the SDP. This is allowed according to the RFC. Support for this differs quite a lot though and some endpoints do not handle it well. This change extends the 'asymmetric_rtp_codec' option to also cover this case. When set to no (the default) the code behaves as chan_sip does - the best codec is selected and we will only ever send that, unless we change what we are sending if the remote side changes. When set to yes we will send media as-is without transcoding if the codec has been negotiated in the SDP. ASTERISK-26996 Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51	9 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	9 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	9 years ago
Sean Bright	d847fe6585	res_agi: Allow configuration of audio format of EAGI pipe ... This change allows the format of the EAGI audio pipe to be changed by setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of the loaded formats. ASTERISK-26124 #close Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd	9 years ago
Jenkins2	57217e4cc2	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages."	9 years ago
Alexei Gradinari	808f299808	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	9 years ago
Joshua Colp	3c36c29c81	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	9 years ago
George Joseph	201346fb7d	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Richard Mudgett	7312cbe803	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	9 years ago
Troy Bowman	fac5115c43	app_queue: Log reason for PAUSEALL/UNPAUSEALL ... We needed the reason for our reporting when agents pause/unpause all of their queues at once. This is a small, simple patch that adds a reason for PAUSEALL and UNPAUSEALL. I have been using it in production for years. ASTERISK-26920 #close Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d	9 years ago
George Joseph	2fe52174de	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff7) (cherry picked from commit 27344675be)	9 years ago
zuul	90634cc184	Merge "rtp_engine: allocate RTP dynamic payloads per session"	9 years ago
Joshua Colp	c1ab8ca74c	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."	9 years ago