1118 Commits (a7d856cd96eb1e1f02fcde6b3e6bef35e1715bd5)

Author	SHA1	Message	Date
Joshua Colp	a7d856cd96	res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. ... This change implements SRV support for the IP based endpoint identifier module. All possible addresses through SRV are looked up and added as matches. If no SRV records are available a fallback to normal host resolution is done. If an IP address is provided then no SRV lookup occurs. This is configured using the "srv_lookups" option on the identify section and defaults to "yes". ASTERISK-26693 Change-Id: I6b641e275bf96629320efa8b479737062aed82ac	10 years ago
Jonathan R. Rose	d96e350256	core/pbx: dialplan show - display filename/line# ... Adds the ability for extensions to be registered to include filename and line number so that dialplan show output can show the filename and line number of a config file responsible for generating a given extension. This only affects config modules that are written to use the new extension registering functions. In this patch, that only includes pbx_config, so extensions registered in extensions.conf and any included extension will be shown in this manner. Extensions registered in this manner will show the filename and line number *instead* of the registrar. ASTERISK-26658 #close Reported by: Jonathan R. Rose Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30	10 years ago
George Joseph	79b09b5f18	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command ... The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	10 years ago
Joshua Colp	faf2194fab	Merge "app_originate: Add option to execute gosub prior to dial"	10 years ago
Richard Mudgett	4b3d3fc741	res_pjsip_outbound_registration.c: Filter redundant statsd reporting. ... Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646	10 years ago
Richard Mudgett	1dfa11b65c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	10 years ago
David Kerr	ddc951060a	app_originate: Add option to execute gosub prior to dial ... Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57	10 years ago
zuul	120a4999f0	Merge "Add support for building RADIUS with radcli"	10 years ago
Mark Michelson	7a8d6bc81b	Bump ARI version to 2.0.0 ... In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665	10 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	10 years ago
Matt Jordan	a72ef38113	res/ari/resource_bridges: Add the ability to manipulate the video source ... In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	10 years ago
Sebastien Duthil	c6d755de11	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	10 years ago
Tzafrir Cohen	97a75e3829	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 #close Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f	10 years ago
Joshua Colp	93a0de1f0e	app_queue: Add mention of 'ABANDON' variable to CHANGES. ... ASTERISK-26558 Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e	10 years ago
Alexander Traud	9ac53877f6	rtp_engine: Allow more than 32 dynamic payload types. ... Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, this change utilizes payload types in the range between 35 and 63 giving room for another 29 payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	10 years ago
Matt Jordan	c30d677333	res/stasis: Add CLI commands for displaying/debugging ARI apps ... This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5	10 years ago
zuul	0ec5abe592	Merge "Remove ASTERISK_REGISTER_FILE."	10 years ago
Joshua Colp	24d0907849	Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads."	10 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	10 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	10 years ago
Badalyan Vyacheslav	01d1d3763f	cdr_radius,cel_radius: Fix old memleak in unload ... - Call "rc_openlog" optional. If you do not call, you will simply NULL instead of a name. - On the one PID can be only one syslog channel. And it can already be run in logger.c - Calling rc_openlog we assigns a new name for the channel syslog. This unexpected behavior for logger.c. Most lesser evil, is to agree on a NULL name syslog if the channel was not launched in logger.c. It also solves the problem of memory leaks. ASTERISK-26455 #close Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd	10 years ago
Joshua Colp	403c4f5833	pjsip: Support dual stack automatically. ... This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	10 years ago
Michael Walton	3e96d491d0	res_rtp_asterisk: Add ice_blacklist option ... Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the form ice_blacklist = <subnet spec>, e.g. ice_blacklist = 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay discovery. This is useful for optimizing the ICE process where a system has multiple host address ranges and/or physical interfaces and certain of them are not expected to be used for RTP. Multiple ice_blacklist configuration lines may be used. If left unconfigured, all discovered host addresses are used, as per previous behavior. Documention in rtp.conf.sample. ASTERISK-26418 #close Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9	10 years ago
Matt Jordan	dd5129d84a	res/ari: Add the Asterisk EID field to outgoing events ... This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d	10 years ago
George Joseph	86e8716952	app_dial: Add the "Q" option to set the cause on unanswered channels ... The "Q" option will set the cause on the unanswered channels when another channel answers. It overrides the default of ANSWERED_ELSEWHERE. NOTE: chan_sip does not support setting the cause on a CANCEL to anything other than ANSWERED_ELSEWHERE. ASTERISK-26446 #close Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47	10 years ago
Corey Farrell	8c5c95ad89	core: Remove ABI effects of LOW_MEMORY. ... This allows asterisk to compiled with LOW_MEMORY to load modules built without LOW_MEMORY. ASTERISK-26398 #close Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d	10 years ago
Tzafrir Cohen	07b95f7c65	sd_notify (systemd status notifications) support ... sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811	10 years ago
Richard Mudgett	ba362822f3	res_pjsip: Add ignore_uri_user_options option. ... This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	10 years ago
zuul	9d54dd04bb	Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."	10 years ago
Aaron An	2a50c29101	res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. ... This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762	10 years ago
Mark Michelson	ac02bbd9a0	ConfBridge: Make some announcements asynchronous. ... Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a	10 years ago
zuul	edaba05fea	Merge "build: Add download capability for external packages"	10 years ago
Alexander Traud	7a12355dbd	chan_sip: Allow Preferred sRTP. ... Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd	10 years ago
George Joseph	6caf6bcdad	build: Add download capability for external packages ... The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/<product_name>. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a	10 years ago
Richard Mudgett	ea929d766d	res_pjsip: Cache global config options. ... We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7	10 years ago
varnav	d2e03c252d	chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820 ... Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth is not supported in IAX2 protocol. Please refer to section 8.6.13 of RFC 5456. But plaintext auth is still supported by Asterisk implementation of IAX2. This support should be dropped. Patch, based on asterisk-dev discussion, adds deprecation warning on startup if 'auth' is set to 'plaintext', changes default values of 'auth' from 'md5, plaintext' to 'md5'. Patch is safe in terms of backwards compatibility, will work even if remote peers have auth=plaintext and we have defaults. auth=plaintext setting will remain deprecated in Asterisk 14 and 15, and IAX2 plaintext support will be removed in Asterisk 16. ASTERISK-22820 #close Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf	10 years ago
George Joseph	534063fd67	res_pjsip: Add contact_user to endpoint ... contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	10 years ago
George Joseph	36b2a40533	autohints: Update CHANGES and extensions.conf.sample ... Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433	10 years ago
Matt Jordan	c315460abb	channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH ... This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b (cherry picked from commit eec60dd773)	10 years ago
Alexei Gradinari	403b63571c	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack ... The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	10 years ago
Alexei Gradinari	9042ad40f2	app_voicemail: Add taskprocessor alert level options. ... On heavy loaded system with IMAP or DB storage, 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. It could happen when the IMAP or DB server dies or is unreachable. It could happen on startup when there are many (thousands) realtime endpoints configured with unsolicited mwi. If the taskprocessor queue reaches the high water level then the alert is triggered and pjsip stops processing new requests until the queue reaches the low water level to clear the alert. This patch adds 2 new 'general' configuration options to tune taskprocessor alert levels: 'tps_queue_high' - Taskprocessor high water alert trigger level. 'tps_queue_low' - Taskprocessor low water clear alert level ASTERISK-26229 #close Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8	10 years ago
zuul	e2bfcb3e58	Merge "codecs: Add iLBC 20."	10 years ago
zuul	7cfd9bf104	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."	10 years ago
zuul	2949e674a5	Merge "Create Asterisk-14: Update CHANGES and UPGRADE files"	10 years ago
Alexander Traud	9be69c1636	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). ... Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957	10 years ago
Alexander Traud	8fb807009f	codecs: Add iLBC 20. ... Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests this. ASTERISK-26218 #close ASTERISK-26221 #close Reported by: Aaron Meriwether Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa	10 years ago
George Joseph	15bf6a87dc	Create Asterisk-14: Update CHANGES and UPGRADE files ... Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc	10 years ago
Richard Mudgett	0d1744e132	chan_dahdi: Add faxdetect_timeout option. ... The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810	10 years ago
Richard Mudgett	e739888d99	res_pjsip: Add fax_detect_timeout endpoint option. ... The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	10 years ago
Alexander Traud	85212f2799	res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. ... Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c	10 years ago