asterisk

Commit Graph

Author	SHA1	Message	Date
Daniel Journo	8182146e85	pjsip: Add option global/regcontext Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62	10 years ago
George Joseph	a41aab477a	pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88	10 years ago
Ward van Wanrooij	d4b10cfb3e	chan_sip: option 'notifyringing' change and doc fix In the sample sip.conf this is written with regard to notifyringing: ;notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) However, this setting changes whether or not any RINGING indications are sent to subscriptions. There is no separate configurable setting that allows to control whether INUSE subscriptions also get sent RINGING. This is however a useful option, to see (using BLF) if somebody else is able to handle an incoming call or if everybody is busy. This patch corrects the documentation for notifyringing (so the documentation matches the functionality) and make notifyringing a tri-state option, by adding the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing = notinuse, only subscriptions that are not INUSE are sent the RINGING signal. The default setting for notifyringing remains set to yes, so the default behaviour is not affected. ASTERISK-25558 Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa	10 years ago
Dade Brandon	ca394161cf	app_amd: Correct maximum_number_of_words functionality & documentation - The maximum_number_of_words was previously documented as being the number of words that when exceeded, would result in the AMD application returning that the audio represents a machine. This was inconsistent with its actual functionality - it was a number of words that when REACHED, would result in determination as a machine. This update corrects the functionality to match the previously documented functionality. This is a backwards incompatible change in configuration file, and has been added to UPGRADE.txt as a result. The sample configuration file and application defaults have been updated so that the default value is now 2, which reflects the same default functionality as previous versions. - Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. - Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093	10 years ago
Mark Michelson	ed13732188	Confbridge: Add a user timeout option This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Corey Farrell	40574a2ea3	chan_sip: Allow websockets to be disabled. This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
Kevin Harwell	691c0e0b31	res_pjsip_outbound_registration: registration stops due to fatal 4xx response During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2	10 years ago
Matt Jordan	2d7a4a3357	main/logger: Add log formatters and JSON structured logs When Asterisk is part of a larger distributed system, log files are often gathered using tools (such as logstash) that prefer to consume information and have it rendered using other tools (such as Kibana) that prefer a structured format, e.g., JSON. This patch adds support for JSON formatted logs by adding support for an optional log format specifier in Asterisk's logging subsystem. By adding a format specifier of '[json]': full => [json]debug,verbose,notice,warning,error Log messages will be output to the 'full' channel in the following format: { "hostname": Hostname or name specified in asterisk.conf "timestamp": Date/Time "identifiers": { "lwp": Thread ID, "callid": Call Identifier } "logmsg": { "location": { "filename": Name of the file that generated the log statement "function": Function that generated the log statement "line": Line number that called the logging function } "level": Log level, e.g., DEBUG, VERBOSE, etc. "message": Actual text of the log message } } ASTERISK-25425 #close Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238	10 years ago
Joshua Colp	309dd2a409	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	a0c31c7a05	res_pjsip: Add rtp_keepalive to sample config file. Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19	10 years ago
Rusty Newton	d02196448b	Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 years ago
Kevin Harwell	93ac45d3bd	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	11 years ago
Joshua Colp	87470f7d81	Merge "tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket."	11 years ago
Alexander Traud	8f3f414d8c	tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket. When a client connects to a server via SSL/TLS, the server commonly utilizes an RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if the server socket is configured with a certificate for either one of those, it would lose its compatibility with RSA-only clients. Now, the server socket can be configured with up to one RSA, ECDSA and DSA key each. For example, if a client is not compatible with SHA-2 hashed certificates like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy clients and ECDSA/SHA-2 for everyone else. ASTERISK-24815 #close Reported by: Alexander Traud patches: tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520) Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a	11 years ago
Joshua Colp	2bbfcfc647	Merge "cdr_adaptive_odbc: Add ability to set character for quoted identifiers."	11 years ago
Richard Mudgett	7103b374ef	chan_dahdi: Improve force_restart_unavailable_chans option description. ASTERISK-25034 Reported by: Richard Mudgett Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30	11 years ago
Rodrigo Ramírez Norambuena	cb79b8ab80	cel_pgsql: Add support for setting schema Add feature to set optional schema parameter on configuration file via 'schema' setting. Fix query to get columns from table while considering schema. If in the database there exists two tables with same name in distinct schemas it will return an error when inserting record. ASTERISK-24967 #close Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c	11 years ago
Rodrigo Ramírez Norambuena	a24ce38e5e	cdr_adaptive_odbc: Add ability to set character for quoted identifiers. Added the ability to set the character to quote identifiers. This allows adding the character at the start and end of table and column names. This setting is configurable for cdr_adaptive_odbc via the quoted_identifiers in configuration file cdr_adaptive_odbc.conf. ASTERISK-25006 Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8	11 years ago
Joshua Colp	ddf9dcaad7	Merge "cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8"	11 years ago
Rodrigo Ramírez Norambuena	8886b724ae	cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8 This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR columns added in Asterisk 1.8. The columns are: * peeraccount * linkedid * sequence When enabled, the columns in the database entry will be populated with the data from the CDR. ASTERISK-24976 #close Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b	11 years ago
Corey Farrell	6b208d8c3b	Sample Configs: Fix syntax error in pjsip.conf The sample pjsip.conf has a few comment lines that are missing the semicolons at the start of the comment, causing the config to fail load. Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352	11 years ago
Richard Mudgett	03c51cf525	chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65	11 years ago
Corey Farrell	55a780d211	Git Conversion: Switch Non-C files to ASTERISK_REGISTER_FILE. This switches files used to generate other sources to use the new ASTERISK_REGISTER_FILE macro. ASTERISK-25026 #close Reported by: Corey Farrell Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740	11 years ago
Joshua Colp	b2153f1f49	Merge "cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version"	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Rodrigo Ramírez Norambuena	358080e86e	cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version Add new column to INSERT new columns added in cdr 1.8 version. The columns are: * peeraccount * linkedid * sequence This feature is configurable in cdr_odbc.conf using a new configuration option, 'newcdrcolumns'. ASTERISK-24976 #close Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127	11 years ago
George Joseph	c6ed681638	res_pjsip: Add global option to limit the maximum time for initial qualifies Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
George Joseph	51886c68dc	pjsip_options: Add qualify_timeout processing and eventing This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Kevin Harwell	66f3fd0028	chan_sip: make progressinband default to no After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ ........ Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b3d01f1fbf	channels/chan_iax2: Add a configuration parameter for call token expiration This patch adds a new configuration parameter, 'calltokenexpiration', that controls how long before an authentication call token is expired. The default maintains the RFC specified 10 seconds. Setting it to a higher value may be useful in lossy networks. Review: https://reviewboard.asterisk.org/r/4588 ASTERISK-24939 #close Reported by: Y Ateya patches: ctoken_configuration.diff submitted by Y Ateya (License 6693) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	016fba12e2	cel_pgsl: Add support for GMT timestamps This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps to be logged in GMT. Review: https://reviewboard.asterisk.org/r/4571/ ASTERISK-23186 #close Reported by: Rodrigo Ramirez Norambuena patches: cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	abf3e40902	dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. This change adds an abstracted core DNS API which resembles the API described here[1]. The API provides a pluggable mechanism for resolvers and also a consistent view for records. Both synchronous and asynchronous queries are supported. This change also adds a res_resolver_unbound module which uses the libunbound library to provide resolution. Unit tests have also been written for all of the above to confirm the API and functionality. ASTERISK-24834 #close Reported by: Matt Jordan ASTERISK-24836 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4474/ Review: https://reviewboard.asterisk.org/r/4512/ [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4c2fc5b811	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ ........ Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	aef7278af6	res_pjsip: Allow configuration of endpoint identifier query order This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ ........ Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	dc752f515b	apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation This patch corrects the documentation for the AMD application. Specifically: * It documents the maximum_word_length option, which limits the maximum allowed length of a single utterance. * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was undocumented. Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. ASTERISK-19470 #close Reported by: Frank DiGennaro ........ Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432920 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d42c6adb1a	Revert - res_pjsip: Allow configuration of endpoint identifier query order Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	1ce529d30e	res_pjsip: allow configuration of endpoint identifier query order It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ ........ Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	55eb8fc068	pjsip: Remove "contact" type from pjsip.conf.sample The "contact" object is not meant to be configured from the pjsip.conf configuration file. It is meant to be created as a result of a registration and stored elsewhere. ASTERISK-24085 #close Reported by: Rusty Newton ........ Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	49161d8df8	res_pjsip_config_wizard: Add ability to auto-create hints. Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => <hint_exten>,1,<hint_application>' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ ........ Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	6a76740b83	HTTP: For httpd server, need option to define server name for security purposes Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ ........ Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e4738a59eb	CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	023fa0f9e8	Add support for the ca_list_path option for PJSIP transports. This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	aa7e06f797	configs/samples/features.conf.sample: Document attended transfer DTMF options The sample config was missing the configuration options for DTMF attended transfer completion scenarios. The configuration options 'atxferabort', 'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the appropriate configuration file. ASTERISK-24678 #close Reported by: Niklas Larsson patches: features.conf.sample.diff uploaded by Niklas Larsson (License 5068) ........ Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	77ee23210d	res_fax: Add T.38 negotiation timeout option This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	91becf952a	app_queue: Update sample conf documenation Updated the queues.conf.sample file to explicitly state which channel queue variables are propagated to. ASTERISK-24267 Reported by: Mitch Claborn ........ Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b79a4a464f	app_confbridge: Add the ability to pass options/command to MixMonitor This patch adds the ability to pass options and a command to MixMontor when recording a conference using ConfBridge. New options are - * record_options: Options to MixMontor, eg: m(), W() etc. * record_command: The command to execute when recording is over. * record_file_timestamp: Append the start time to the file name. These options can also be used with the CONFBRIDGE function, e.g., Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME})) Review: https://reviewboard.asterisk.org/r/4023 ASTERISK-24351 #close Reported by: Gareth Palmer patches: record_command-428838.patch uploaded by Gareth Palmer (License 5169) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	39b54a21dc	res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios res_pjsip_config_wizard ------------------ * This is a new module that adds streamlined configuration capability for chan_pjsip. It's targetted at users who have lots of basic configuration scenarios like 'phone' or 'agent' or 'trunk'. Additional information can be found in the sample configuration file at config/samples/pjsip_wizard.conf.sample. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4190/ ........ Merged revisions 429592 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1106e8fd0f	main/stasis: Allow subscriptions to use a threadpool for message delivery Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Olle Johansson	d663e045f5	sip.conf.sample - note that media_address does not change listen address, just the SDP git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7f8b7ace72	res_pjsip_sdp_rtp: Add support for optimistic SRTP. Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2e750db120	Allow for transferer to retry when dialing an invalid extension. This allows for a configurable number of attempts for a transferer to dial an extension to transfer the call to. For Asterisk 13, the default values are such that upgrading between versions will not cause a behaivour change. For trunk, though, the defaults will be changed to be more user-friendly. Review: https://reviewboard.asterisk.org/r/4167 ........ Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d0523b4b3c	chan_sip: Add support for setting DTLS configuration in the general section. Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	df2090b931	Documentation: Revise explanation of cdr.conf option 'Unanswered' ASTERISK-24279 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4109/ ........ Merged revisions 427901 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	69f29e627f	Make the disable_tcp_switch PJSIP system object enabled by default. Testing has shown repeatedly that PJSIP's default behavior of switching automatically to TCP for large messages can cause issues. The most common issues are that devices that we are communicating with do not handle the switch to TCP gracefully, thus causing situations such as broken calls or broken subscriptions. Now, in order to have this behavior happen, you must opt into it. The sample file has been updated to warn that enabling the TCP switch behavior may cause issues for you, so use at your own risk. ........ Merged revisions 427334 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d159885e50	res_pjsip_outbound_registration: Add virtual line support. Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. Review: https://reviewboard.asterisk.org/r/2964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	33f0251b6c	res_pjsip: Add disable_tcp_switch option. When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In some circumstances (on some networks), this can cause some issues with messages not getting sent to the correct destination - and can also cause connections to get dropped due to quirks in pjproject deciding to terminate TCP connections with no messages. While fixing the routing/messaging issues is important, having a configuration option in Asterisk that tells pjproject to not switch over to TCP would be useful. That way, if some glitch is discovered on some other network/site, we can at least disable the behavior until a fix is put into place. AFS-197 #close Review: https://reviewboard.asterisk.org/r/4137/ ........ Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427130 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ac091d4184	chan_pjsip: Add support for passing hold and unhold requests through. This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	28173ddf05	pjsip: clarify tls cert and key file usage A question arose as to whether a .pem file could be provided in place of the .crt and .key files in a PJSIP TLS configuration. I tested this and discovered that although a cert will be read from the pem file, a key will not, and thus the priv_key_file entry is still required. This update to the fine documentation clarifies the option usage. AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ Reported by: John Bigelow ........ Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426930 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	1fe22c411d	ASTERISK-23512, correct inaccurate comment in manager.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	684b8762a9	ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b8f687f27c	Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules AST-1432 #close Reported by: John Bigelow ........ Merged revisions 425905 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0d0e38a0e1	res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c6837c236f	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc595f7353	res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b1f8eba178	PJSIP: Restore functional default for callerid_privacy The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424427 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2b0777c017	res_pjsip: Make transport cipher option accept a comma separated list of cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424394 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	aa5458d6ab	res_pjsip: document use of rewrite_contact in sample conf Without setting rewrite_contact, an invite to an endpoint behind NAT will not reach it - unless the endpoint itself uses STUN or TURN to discover it's public URI. Thus, the use of this should be in the sample documentation. Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	adba2a8d7f	res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424291 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4d2c7c23f8	PJSIP: Handle defaults properly This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	d172d84fe1	musiconhold: Add preferchannelclass=no option to prefer app class. The new option 'preferchannelclass' is added to musiconhold.conf. If yes (the default) the CHANNEL(musicclass) is preferred when choosing the hold music. If it is no, the class suggested by the application that calls the MoH (e.g. the Queue() app) gets preferred (new behaviour). This way you set a different hold-music from the Queue-music by setting both the CHANNEL(musicclass) and the queue-context musicclass. ASTERISK-24276 #close Reported by: Kristian Høgh Patches: app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639) Review: https://reviewboard.asterisk.org/r/4010/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a62fedf0cb	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423069 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9c1f34c7e9	musiconhold: Add sort=randstart, and deprecate old stuff. - adds sort=randstart (next to sort=, sort=random, sort=alpha) - combines duplicate moh option parsing code into a single function - adds deprecationwarnings for application=r to sort randomly - adds deprecationwarnings for random=yes to sort randomly - removes invisible code that was supposed to stay until 1.8 The sort=randstart works like sort=alpha, except we start at a random position. Review: https://reviewboard.asterisk.org/r/3991/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	baf99dffac	Add note about configuring list_items on a single line. ........ Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	5ad0edacb6	Add sample configuration for resource lists. On review /r/3977, it was recommended to note in the sample configuration about the size limitation for resource lists. However, since there was no section in the sample configuration at all for resource list subscriptions, I decided to make a separate commit where I have added the necessary sample configuration as well as the size limitation warning. ........ Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	add46fd27c	app_queue: Add RealTime support for queue rules This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) ........ Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	5ce4ad8031	app_voicemail: Add the ability to specify multiple email addresses. ASTERISK-24045 Reported by: Jacob Barber Review: https://reviewboard.asterisk.org/r/3833/ ........ Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fc0fecb476	configs: Move sample config files into a subdirectory of configs This moves all samples configs from configs/ to configs/samples. This allows for additional sets of sample configuration files to be added in the future. Review: https://reviewboard.asterisk.org/r/3804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

... 5 6 7 8 9

429 Commits (01cbaffbdfadff30aad43541b505ed37e02d7a2b)