asterisk

Commit Graph

Author	SHA1	Message	Date
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
Richard Mudgett	8baedc2204	res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header This patch adds regular expression support to make the identify section's match_header option more useful when attempting to match complex headers like the 'To' or 'From' headers. The 'From' header has variable components such as the tag parameter that you cannot predict. To specify a regular expression put slashes around the regular expression in place of the header value. [identify-alice] type=identify endpoint=alice match_header=From: /<sip:alice@127\\.0\\.0\\.1>/ * Added regex support to match_header so you could match a 'To' header among other complex headers. Fixed reported crashes when trying to match special headers like 'Contact'. The identify section's match_header method used code that assumed you were matching a generic header. Any other type of header could cause a crash if the header structure variant did not match the generic header enough. * Made use code that will work for any header type instead of code specific to generic headers. Other fixes while in the area: * Made check all headers of the requested name. * Added some more sanity checks to the configured identify matching options when applying the configuration. ASTERISK-27548 Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1	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
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
William McCall	9ff4779f03	app_confbridge: Add talking indicator for ConfBridgeList AMI response When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	7 years ago
George Joseph	2b50b31b68	Merge "app_sendtext: Enhance SendText to support Enhanced Messaging" into 13	7 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	7 years ago
George Joseph	3c2249fd37	app_sendtext: Enhance SendText to support Enhanced Messaging SendText now accepts new channel variables that can be used to override the To and From display names and set the Content-Type of a message. Since you can now set Content-Type, other text/* content types are now valid. Change-Id: I648b4574478119f95de09d9f08e9595831b02830	7 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	7 years ago
Nathan Bruning	6a4afe09ce	res_pjsip_notify.c: enable in-dialog NOTIFY This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29	7 years ago
Alexander Traud	83353997f4	BuildSystem: Add support for building RADIUS with radcli. Radcli is yet another RADIUS client library, generally compatible with freeradius and radiusclient-ng. This commit adds autoconf option for detecting it as well and changes cdr_radius and cel_radius to use its header file in that case. ASTERISK-26540 Reported by: Tzafrir Cohen Change-Id: Icc056d476b7acf481309219e9abdca416866c6ec	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
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
George Joseph	5947fd148b	AST-2018-005: res_pjsip_transport_management: Move to core Since res_pjsip_transport_management provides several attack mitigation features, its functionality moved to res_pjsip and this module has been removed. This way the features will always be available if res_pjsip is loaded. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d	7 years ago
Richard Mudgett	1ff580bb25	app_confbridge: ConfbridgeList event has standard channel shapshot headers. * Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977	7 years ago
Richard Mudgett	0cf7a9e0ca	app_confbridge: Add the Muted header to ConfbridgeJoin AMI event. ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34	7 years ago
George Joseph	7debdd285c	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	7 years ago
Corey Farrell	698b28e9c7	res_pjsip_registrar_expire: Mark module deprecated. The functionality of this module was already moved to res_pjsip_registrar, mark it deprecated and add message to CHANGES. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813	7 years ago
krells	d5bcbd460e	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	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	8 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	e3bd95f55c	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
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
Igor Goncharovskiy	8436f1d35a	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	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
Joshua Colp	7215d07ca2	Merge "features, manager : Add CancelAtxfer AMI action" into 13	8 years ago
Thomas Sevestre	6d3ee9fb93	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	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
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
Jenkins2	fa50a3def9	Merge "res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif" into 13	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
George Joseph	363d61ef58	configure: Add --with-download-cache option 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'. Change-Id: I5c3cf15ee61e8fe191b52732303e969854f8d861	8 years ago
Niklas Larsson	9a09f7dd5d	app_queue: Add priority to AMI QueueStatus Add priority to callers in AMI QueueStatus response ASTERISK-27092 #close Change-Id: I8d1f737a72c7c38f4cfe1a4ee3ecc0a4f85bd199	8 years ago
Joshua Colp	114602f434	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
Torrey Searle	423d01cf16	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
Joshua Colp	24bb5a8908	core: Add VP9 passthrough support. This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	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	bbe68f139d	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
Jenkins2	997c11235e	Merge "app_voicemail: IMAP connection control" into 13	8 years ago
Torrey Searle	9fbc34d2bd	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	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
Alexei Gradinari	a6e4899612	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
Joshua Colp	bc51d4324a	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13	8 years ago
Alexei Gradinari	5520b6c201	CHANGES: correct version for a new option 'refer_blind_progress' Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97	8 years ago
Joshua Colp	996a4791ff	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	8 years ago
Jenkins2	812f5b51cb	Merge "res_pjsip: Add support for returning only reachable contacts and use it." into 13	8 years ago
Joshua Colp	746c2c5745	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	8 years ago
Jenkins2	3e8eea0325	Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13	8 years ago

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1108 Commits (c4b979a84172cf14d0c00a24906b8639fd798327)