asterisk

Commit Graph

Author	SHA1	Message	Date
Walter Doekes	cea6cad343	chan_sip: Clarify in sample docs how directmediapermit/-acl should be used It said "restrict [...] which peers should be able to pass [audio] to each other". However, these settings are not global (for which you would expect signaling IPs to be checked). These settings are available per peer only, and the IPs being checked, are the RTP IPs. Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660	5 years ago
Sean Bright	b6ebdabd9c	http: Add ability to disable /httpstatus URI Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	5 years ago
Sean Bright	14de6b9ba7	func_odbc.conf.sample: Add example lookup Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	5 years ago
Sean Bright	49bb7d85cd	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	5 years ago
Joshua C. Colp	ca35de4282	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
Joshua Colp	d775bc5add	chan_pjsip: Add support for passing hold and unhold requests through. This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ Change-Id: Ib6294e906e577e1a4245cb1f058d3976ff484c52	6 years ago
Joshua Colp	1ea9bad34d	res_rtp_asterisk: Add support for DTLS packet fragmentation. This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	6 years ago
George Joseph	90fe830a77	res_rtp_asterisk: Add ability to propose local address in ICE You can now add the "include_local_address" flag to an entry in rtp.conf "[ice_host_candidates]" to include both the advertized address and the local address in ICE negotiation: [ice_host_candidates] 192.168.1.1 = 1.2.3.4,include_local_address This causes both 192.168.1.1 and 1.2.3.4 to be advertized. Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db	6 years ago
Dan Cropp	00588226c6	res_pjsip: Added a norefersub configuration setting Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	6 years ago
Friendly Automation	4f6daa0aff	Merge "chan_pjsip: add a flag to ignore 183 responses if no SDP present" into 13	6 years ago
Torrey Searle	cbc704c5ec	chan_pjsip: add a flag to ignore 183 responses if no SDP present chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	6 years ago
Sean Bright	9a4027ad41	samples: Fix comment typo in pjsip.conf.sample Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857	6 years ago
Joshua Colp	1f3d6e1ebf	basic-pbx: Update configuration to work with current modules. The res_pjsip_websocket module requires the res_http_websocket module so ensure it is loaded. As well the res_pjsip_notify module needs the pjsip_notify.conf configuration file so ensure it is installed. ASTERISK-28272 Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd	6 years ago
George Joseph	bae3fd04c1	taskprocessor: Enable subsystems and overload by subsystem To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	6 years ago
Joshua Colp	a4d930c2ed	res_odbc: Add basic query logging. When Asterisk is connected and used with a database the response time of the database can cause problems in Asterisk if it is long. Normally the only way to see this problem would be to retrieve a backtrace from Asterisk and examine where things are blocked, or examine the database to see if there is any indication of a problem. This change adds some basic query logging to make it easier to investigate such a problem. When logging is enabled res_odbc will now keep track of the number of queries executed, as well as the query that has taken the longest time to execute. There is also an option which will cause a WARNING message to be output if a query takes longer than a configurable amount of time to execute. This makes it easier and clearer for users that their database may be experiencing a problem that could impact Asterisk. ASTERISK-28277 Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6	6 years ago
Kevin Harwell	62faecd60d	codecs.conf.sample: update codec opus docs The option value "sdp" for some of the settings was removed a while back, however the sample conf was not updated. This patch removes any wording with regards to the old "sdp" option value, and adjusts the defaults to what they are now. ASTERISK-28263 Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445	6 years ago
Friendly Automation	9514829403	Merge "app_voicemail: Add Mailbox Aliases" into 13	6 years ago
George Joseph	d0eccac3ff	app_voicemail: Add Mailbox Aliases You can now define an "aliases" context in voicemail.conf whose entries point to actual mailboxes. These can be used anywhere the mailbox is specified. Example: [general] aliasescontext = myaliases [default] 1234 = yadayada [myaliases] 4321@devices = 1234@default Now you can use 4321@devices to refer to the 1234@default mailbox. This can be useful to provide channel drivers with constant mailbox specifications such as <extension>@devices leaving app_voicemail to control exactly which mailbox the alias points to. Now, only voicemail has to be reloaded to make changes instead of individual channel drivers which are usually more expensive to reload. Change-Id: I395b9205c91523a334fe971be0d1de4522067b04	6 years ago
Alexei Gradinari	4a8564cafa	res_pjsip: add option to disable ContactStatus event when contact is updated This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to have the same performance benefits as version 16. By default old behavior, i.e. the ContactStatus event will be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	6 years ago
George Joseph	363d19326b	Revert "app_voicemail: Remove need to subscribe to stasis" This reverts commit `1843b0e2b5`. That commit closed a long standing hole which allowed subscriptions to mailboxes that weren't configured in voicemail.conf. This caused an issue with FreePBX which depdended on that behavior. The commit is being reverted until FreePBX can handle the new behavior. ASTERISK-28151 Reported by: Ronald Raikes Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15	7 years ago
Alexei Gradinari	bfe3821800	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
Sean Bright	78aee0f6a2	samples: PARKINGSLOT -> PARKING_SPACE in parking sample config PARKINGSLOT was deprecated in Asterisk 12 but the sample config was not updated to reflect that. Change-Id: I3e087c19d9ee587094fa5304102d8084a79c2b3c	7 years ago
Ben Ford	aa31657e28	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
George Joseph	1843b0e2b5	app_voicemail: Remove need to subscribe to stasis app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Matthew Fredrickson	adb3195697	sample_configs: noload res_hep.so by default Change disables loading of res_hep.so in default installation. Loading res_hep has a performance impact whether it's used or not. This disables loading of it in sample config files. Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0 (cherry picked from commit `c8bacd45f1`)	7 years ago
Richard Mudgett	0a7dab8904	pbx_dundi: Update sample config documentation. Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849	7 years ago
Corey Farrell	dc786aa576	Sample configs: Fix pjsip.conf syntax error. It is valid for a config file to be empty or contain only comments, but not valid for a config value to be set when no uncommented context exists. This caused an error to be loged numerous times during start when loading the default pjsip.conf. Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6	7 years ago
Richard Mudgett	75131c9e1c	pjsip_wizard.conf.sample: Update remote_hosts description. Remove the note that SRV records are not supported as that is no longer true. ASTERISK-27993 Change-Id: Id0dd6ef40e52702be9727a2b6122216cb00bb4ca	7 years ago
Richard Mudgett	0ade9df3b6	res_pjsip: Update endpoint transport option documentation. Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52	7 years ago
George Joseph	3470409dd6	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	7 years ago
Joshua Colp	62859ad526	pjsip: Clarify certificate configuration for Websocket. The Websocket transport uses the built-in HTTP server. As a result the TLS configuration is done in http.conf and not in pjsip.conf. This change adds a warning if this is configured in pjsip.conf and also clarifies in the sample configuration file. Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9	7 years ago
George Joseph	06966e91fe	res_pjsip_session: Add ability to accept multiple sdp answers pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	7 years ago
George Joseph	acfdfcd19e	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
George Joseph	373e7e3fb0	channel.c: Allow generic plc then channel formats are equal If the two formats on a channel are equal, we don't transcode and since the generic plc needs slin to work, it doesn't get invoked. * A new configuration option "genericplc_on_equal_codecs" was added to the "plc" section of codecs.conf to allow generic packet loss concealment even if no transcoding was originally needed. Transcoding via SLIN is forced in this case. ASTERISK-27743 Change-Id: I0577026a179dea34232e63123254b4e0508378f4	7 years ago
Jenkins2	2961dd6c6d	Merge "core: Fix handling of maximum length lines in config files." into 13	7 years ago
Richard Mudgett	104468ad3a	pjproject: Add cache_pools debugging option. The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	7 years ago
Corey Farrell	caad0c09cd	core: Fix handling of maximum length lines in config files. When a line is the maximum length "\n" is found at sizeof(buf) - 2 since the last character is actually the null terminator. In addition if a line was exactly 8190 plus a multiple of 8192 characters long the config parser would skip the following line. Additionally fix comment in voicemail.conf sample config. It previously stated that emailbody can only contain up to 512 characters which is always wrong. The buffer is normally 8192 characters unless LOW_MEMORY is enabled then it is 512 characters. The updated comment states that the line can be up to 8190 or 510 characters since the line feed and NULL terminator each use a character. ASTERISK-26688 #close Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015	7 years ago
Richard Mudgett	4a337b1a76	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	7 years ago
ghjm	0b399013c6	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	7 years ago
Richard Mudgett	f35960d55b	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	7 years ago
Richard Mudgett	2e09ed3b18	res_pjsip.c: Update the endpoint identification documentation. * Endpoint identify_by documentation. * IP/Header endpoint identifier documentation. Change-Id: Id92f00b495acca7be945daf749d2abd7f76a0b5a	8 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Joshua Colp	7f2df9e277	confbridge: Clarify mute sound documentation. The mute/unmute sounds are only played when the action is initiated using the DTMF menu. ASTERISK-24756 Change-Id: I55b3dd5bc166096bf5e2f547ddd0ce355f36e3dc	8 years ago
Jenkins2	ccb563d357	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams." into 13	8 years ago
Richard Mudgett	61e81338d9	res_rtp_asterisk.c: Disable packet flood detection for video streams. We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
Sean Bright	a1fcb7b5a6	configs: Comment out and change IP of iax.conf [demo] This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100	8 years ago
Alexander Traud	e819cf7826	res_rtp_asterisk: Correct default in sample configuration file. With Asterisk 12 (commit `866d968`), the default of "icesupport" changed to - "yes" in the module "res_rtp_asterisk" and - "no" in the module "chan_sip". The latter was reflected in the sample configuration file for "sip.conf". The former did not make it into "rtp.conf.sample". ASTERISK-20643 Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36	8 years ago
Richard Mudgett	8dd9a79e6e	features.conf.sample: Clarify ActivatedBy documentation wording. Change-Id: Id2899331fe05d1909a862ea879742879d086bc64	8 years ago
George Joseph	062a4390ac	ast_coredumper: Add ability to use directory other than /tmp The OUTPUTDIR environment variable can now be set either in the environment itself or in ast_debug_tools.conf. If set, it's used for all work products instead of /tmp. Also added the --tarball-config option that includes the contents of /etc/asterisk when either --tarball-coredumps or --tarball-results are used. Change-Id: I66b2553319df61caea5b313d084f51978f730b4c	8 years ago
Richard Mudgett	507d9b5f9e	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

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1933 Commits (40ef98e9593da68ed9fdb3b1b176d347cb7001ef)