288 Commits (9e19d98f45741bfebb62cbd69cfe1f1b557ea2d5)

Author	SHA1	Message	Date
Sean Bright	526917bf61	modules.conf: Fix differing usage of assignment operators. ... ASTERISK-24434 #close Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db	5 years ago
Jaco Kroon	d847f7e8f8	func_odbc: Introduce minargs config and expose ARGC in addition to ARGn. ... minargs enables enforcing of minimum count of arguments to pass to func_odbc, so if you're unconditionally using ARG1 through ARG4 then this should be set to 4. func_odbc will generate an error in this case, so for example [FOO] minargs = 4 and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a potentially leaked ARG4 from Gosub(). ARGC is needed if you're using optional argument, to verify whether or not an argument has been passed, else it's possible to use a leaked ARGn from Gosub (app_stack). So now you can safely do ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Sebastien Duthil	435d68be97	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	5 years ago
Alexander Traud	c98de279b2	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
George Joseph	10c2e1749b	chan_iax2.c: Require secret and auth method if encryption is enabled ... If there's no secret specified for an iax2 peer and there's no secret specified in the dial string, Asterisk will crash if the auth method requested by the peer is MD5 or plaintext. You also couldn't specify a default auth method in the [general] section of iax.conf so if you don't have static peers defined and just use the dial string, Asterisk will still crash even if you have a secret specified in the dial string. * Added logic to iax2_call() and authenticate_reply() to print a warning and hanhup the call if encryption is requested and there's no secret or auth method. This prevents the crash. * Added the ability to specify a default "auth" in the [general] section of iax.conf. ASTERISK-29624 Reported by: N A Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254	5 years ago
lvl	8d2558209b	Introduce astcachedir, to be used for temporary bucket files ... As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, but leaves it at /tmp by default, to ensure backwards compatibility. A future commit that only targets master could change the default location to something more sensible such as /var/tmp/asterisk. At that point, the cachedir could be created and cleaned up during uninstall by the Makefile script. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	5 years ago
Alexander Traud	d1a78e047d	modules.conf: Align the comments for more conclusiveness. ... Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306	5 years ago
Dovid Bender	5046e1fb06	func_curl.c: Allow user to set what return codes constitute a failure. ... Currently any response from res_curl where we get an answer from the web server, regardless of what the response is (404, 403 etc.) Asterisk currently treats it as a success. This patch allows you to set which codes should be considered as a failure by Asterisk. If say we set failurecodes=404,403 then when using curl in realtime if a server gives a 404 error Asterisk will try to failover to the next option set in extconfig.conf ASTERISK-28825 Reported by: Dovid Bender Code by: Gobinda Paul Change-Id: I94443e508343e0a3e535e51ea6e0562767639987	5 years ago
Sean Bright	e326b133dc	features.conf.sample: Sample sound files incorrectly quoted ... ASTERISK-29136 #close Change-Id: I3186536d65a50014c8da4780c9224919caa81440	5 years ago
Andrew Siplas	606bd35060	logger.conf.sample: add missing comment mark ... Add missing comment mark from stock configuration. ASTERISK-29123 #close Change-Id: I4f94eb4544166bca8af4c17fd11edee3c6980620	5 years ago
Ben Ford	21ab0a450b	res_stir_shaken: Add stir_shaken option and general improvements. ... Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	5 years ago
Ben Ford	d979bdf87a	res_stir_shaken: Add outbound INVITE support. ... Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is sent, the caller ID will be checked to see if there is a certificate that corresponds to it. If so, that information will be retrieved and an Identity header will be added to the SIP message. The format is: header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken Header, payload, and signature are all BASE64 encoded. The public key URL is retrieved from the certificate. Currently the algorithm and ppt are ES256 and shaken, respectively. This message is signed and can be used for verification on the receiving end. Two new configuration options have been added to the certificate object: attestation and origid. The attestation is required and must be A, B, or C. origid is the origination identifier. A new utility function has been added as well that takes a string, allocates space, BASE64 encodes it, then returns it, eliminating the need to calculate the size yourself. Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4	5 years ago
Ben Ford	035b463c93	res_stir_shaken: Added dialplan function and API call. ... Adds the "STIR_SHAKEN" dialplan function and an API call to add a STIR_SHAKEN verification result to a channel. This information will be held in a datastore on the channel that can later be queried through the "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results including identity, attestation, and verify_result. Here are some examples: STIR_SHAKEN(count) STIR_SHAKEN(0, identity) STIR_SHAKEN(1, attestation) STIR_SHAKEN(2, verify_result) Getting the count can be used to iterate through the results and pull information by specifying the index and the field you want to retrieve. Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82	5 years ago
George Joseph	3b0a53f257	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
Sean Bright	fccf360fcb	res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs ... Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e	5 years ago
Alexander Traud	afc5cf8d67	samples: Fix keep_alive_interval default in pjsip.conf. ... Since ASTERISK_27978 the default is not off but 90 seconds. That change happened because ASTERISK_27347 disabled the keep-alives in the bundled PJProject and Asterisk should behave the same as before. Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46	6 years ago
George Joseph	d28a44c33d	logger.c: Added a new log formatter called "plain" ... Added a new log formatter called "plain" that always prints file, function and line number if available (even for verbose messages) and never prints color control characters. It also doesn't apply any special formatting for verbose messages. Most suitable for file output but can be used for other channels as well. You use it in logger.conf like so: debug => [plain]debug console => [plain]error,warning,debug,notice,pjsip_history messages => [plain]warning,error,verbose Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d	6 years ago
sungtae kim	e34da79c60	res_pjsip.c: Added disable_rport option for pjsip.conf ... Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	6 years ago
George Joseph	6a0c472374	Scope Tracing: A new facility for tracing scope enter/exit ... What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027	6 years ago
Joshua C. Colp	62183bc777	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	6 years ago
Jaco Kroon	351b2be00a	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. ... A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Sean Bright	268b0a3085	func_odbc.conf.sample: Clarify sample documentation ... ASTERISK-20325 #close Change-Id: I06cb9b467b0fd06c8af2a5aee049f872c09cc4b6	6 years ago
Jared Smith	a9abe55e0b	indications.conf.sample: Add indication tones for Indonesia ... These tones come from http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf ASTERISK-23407 Change-Id: I48e2285f1e5bb29b3335f762006f66c423d0fcb8	6 years ago
Walter Doekes	35b4f1686b	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	6 years ago
George Joseph	d6574cb7c7	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Sean Bright	a2a4e1026c	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	6 years ago
Sean Bright	a7aaca9eaa	func_odbc.conf.sample: Add example lookup ... Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	6 years ago
Sean Bright	f8b0c2c933	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	6 years ago
Joshua C. Colp	5622df0a94	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
Sean Bright	9f304170f6	res_musiconhold: Add new 'playlist' mode ... Allow the list of files to be played to be provided explicitly in the music class's configuration. The primary driver for this change is to allow URLs to be used for MoH. Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa	6 years ago
Joshua Colp	82789aafd6	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	7 years ago
Kirsty Tyerman	a1c84709b8	pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi ... ASTERISK-28234 Reported-by: Kirsty Tyerman Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1	7 years ago
George Joseph	79b15d0b30	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	7 years ago
Joshua Colp	d861ebdca8	app_confbridge: Add "all" variants of REMB behavior. ... When producing a combined REMB value the normal behavior is to have a REMB value which is unique for each sender based on all of their receivers. This can result in one sender having low bitrate while all the rest are high. This change adds "all" variants which produces a bridge level REMB value instead. All REMB reports are combined together into a single REMB value that is the same for each sender. ASTERISK-28401 Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c	7 years ago
Dan Cropp	eca8c440d2	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	7 years ago
Torrey Searle	4ca41caf38	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	7 years ago
Sean Bright	90f06ad169	samples: Fix comment typo in pjsip.conf.sample ... Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857	7 years ago
George Joseph	2f8def1453	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	7 years ago
Joshua Colp	44254d548b	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	7 years ago
Kevin Harwell	f6452f9656	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	7 years ago
George Joseph	dbef559e0b	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	7 years ago
Alexei Gradinari	7f22c9f4b7	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	7 years ago
George Joseph	a2a8a41db3	Revert "app_voicemail: Remove need to subscribe to stasis" ... This reverts commit 29115e2384. 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	5cbe77cc46	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
Richard Mudgett	11427a9665	modules.conf.sample: Update preload usage documentation. ... Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90	7 years ago
Sean Bright	495a1e7e5c	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	67e1e49e08	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	29115e2384	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	4188e7d6dd	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)	8 years ago
Richard Mudgett	a7ace944ae	pbx_dundi: Update sample config documentation. ... Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849	8 years ago