1914 Commits (13.24)

Author	SHA1	Message	Date
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	7 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	7 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	7 years ago
Jenkins2	ccb563d357	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams." into 13	7 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	7 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	7 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
Joshua Colp	7385d1e017	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
Jenkins2	5a8c148dcf	Merge "res_pjsip_registrar.c: Update remove_existing AOR contact handling." into 13	8 years ago
Richard Mudgett	d388c18abf	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
Sean Bright	6b16fa12c8	res_config_sqlite: Don't enable SQLite CDRs when running 'make samples' ... Change-Id: I65a5190b2732b2246d67472db70dd37db64ddad4	8 years ago
Jenkins2	b6e1b13de4	Merge "res_pjsip: Filter out non SIP(S) requests" into 13	8 years ago
George Joseph	63900374fa	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
George Joseph	ed2a4ee81e	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
Corey Farrell	04ee3eb774	AST-2017-006: Fix app_minivm application MinivmNotify command injection ... An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	8 years ago
Andrey Egorov	54e3ac402f	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
Jenkins2	a5e73ad1f8	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO." into 13	8 years ago
Corey Farrell	df49ad2528	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. ... This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	8 years ago
Sean Bright	6d0ff310c6	basic-pbx: Remove res_pjsip_multihomed from sample config ... ASTERISK-27127 #close Reported by: HZMI8gkCvPpom0tM Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789	8 years ago
George Joseph	4e555437dc	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
George Joseph	40490768cc	Merge "chan_pjsip: Fix ability to send UPDATE on COLP" into 13	8 years ago
George Joseph	6bd7c0f37c	chan_pjsip: Fix ability to send UPDATE on COLP ... When connected_line_method is "invite", we're supposed to determine if the client can support UPDATE and if it can, send UPDATE instead of INVITE to avoid the SDP renegotiation. Not only was pjproject not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing that invite_tsx wasn't NULL which isn't always the case. * Updated chan_pjsip/update_connected_line_information to drop the requirement that invite_tsx isn't NULL. * Submitted patch to pjproject sip_inv.c that sets the PJSIP_INV_SUPPORT_UPDATE flag correctly. * Updated pjsip.conf.sample to clarify what happens when "invite" is specified. ASTERISK-27095 Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560	8 years ago
Jenkins2	997c11235e	Merge "app_voicemail: IMAP connection control" into 13	8 years ago
Rodrigo Ramírez Norambuena	cecf6540dc	cdr: fix mistake spelling of a word for Unanswered. ... Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	8 years ago
Alexei Gradinari	8f356192d1	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
Jenkins2	707e0e62e6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" into 13	8 years ago