asterisk

Commit Graph

Author	SHA1	Message	Date
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
Mark Michelson	36a9bd994b	Merge "CHANGES: Add missing spaces."	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
Rodrigo Ramírez Norambuena	542bfee881	CHANGES: Add missing spaces. Change-Id: I534ea0f22759e3633585dfa9b145b4a284efe67f	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
Rodrigo Ramírez Norambuena	2a36bb5d9a	CHANGES remove tab space Change-Id: I6b43e43474bf6fb77b8227eadb036036f8e90521	11 years ago
Matt Jordan	bb347fa594	Merge topic 'ASTERISK-24863' * changes: res_pjsip: Add global option to limit the maximum time for initial qualifies pjsip_options: Add qualify_timeout processing and eventing res_pjsip: Refactor endpt_send_request to include transaction timeout	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
Joshua Colp	a3cec44a0a	res_pjsip: Add external PJSIP resolver implementation using core DNS API. This change adds the following: 1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked. 2. Unit tests for the query set implementation. 3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups. For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A, with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit transport has been provided. Configured transports on the system are taken into account to eliminate resolved addresses which have no hope of completing. ASTERISK-24947 #close Reported by: Joshua Colp Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e	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
Kevin Harwell	520b9f2174	res_pjsip: add CLI command to show global and system configuration Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings ASTERISK-24918 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/4597/ ........ Merged revisions 434527 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434528 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
Matthew Jordan	c2f50ba6f4	ARI: Add the ability to intercept hold and raise an event For some applications - such as SLA - a phone pressing hold should not behave in the fashion that the Asterisk core would like it to. Instead, the hold action has some application specific behaviour associated with it - such as disconnecting the channel that initiated the hold; only playing MoH to channels in the bridge if the channels are of a particular type, etc. One way of accomplishing this is to use a framehook to intercept the hold/unhold frames, raise an event, and eat the frame. Tasty. This patch accomplishes that using a new dialplan function, HOLD_INTERCEPT. In addition, some general cleanup of raising hold/unhold Stasis messages was done, including removing some RAII_VAR usage. Review: https://reviewboard.asterisk.org/r/4549/ ASTERISK-24922 #close ........ Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 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	ac1214d9d4	apps/app_sms: Add an option to prevent SMS content from being logged In some countries, privacy laws specify that SMS content cannot be saved by a provider. This patch adds a new option to the SMS application, 'n', which prevents the SMS content from being written to the SMS log. ASTERISK-22591 #close Reported by: Jan Juergens patches: DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 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
Matthew Jordan	29f66b0429	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. New Feature A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. Bug fixes In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 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	12 years ago
Mark Michelson	1111944afb	Change PJProject version requirement for ca_list_path transport option in CHANGES file. ........ Merged revisions 430716 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Richard Mudgett	52a7cdb101	AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Mark Michelson	7f836c1c15	Add the ability to continue and originate using priority labels. With this patch, the following two ARI commands POST /channels POST /channels/{id}/continue Accept a new parameter, label, that can be used to continue to or originate to a priority label in the dialplan. Because this is adding a new parameter to ARI commands, the API version of ARI has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 #close Reported by Nir Simionovich Review: https://reviewboard.asterisk.org/r/4285 ........ Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	fb3c8e3424	outbound_registration: Add 'pjsip send register' and update 'send unregister' The current behavior of 'pjsip send unregister' is to send the unregister (REGISTER with 0 exp) but let the next scheduled register proceed normally. I don't think that's a good idea. If you unregister, it should stay unregistered until you decide to start registrations again. So this patch just adds a cancel_registration call to the current unregister_task to cancel the timer. Of course, now you need a way to start registration again so I've added a 'pjsip send register' command that unregisters and cancels any existing registration (the same as send unregister), then sends an immediate registration and starts the timer back up again. Both changes also ripple to AMI. There's a new PJSIPRegister command. There's no harm in calling either command repeatedly. They don't care about the actual state. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4301/ ........ Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e0bd2ca104	pjsip: Document addition of 'PJSIP_AOR' and 'PJSIP_CONTACT' in CHANGES file. ........ Merged revisions 430181 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Richard Mudgett	eacbb4ceb5	chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode. For the featdmf signaling mode the incoming MF Caller-ID information is formatted as follows: ${CALLERID(ani2)}${CALLERID(ani)}#${EXTEN}# Rather than discarding the ani2 digits, populate the CALLERID(ani2) value with what is received instead. AST-1368 #close Reported by: Denis Martinez Patches: extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett ........ Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429784 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Kevin Harwell	63d3f0af95	ARI/AMI: Include language in standard channel snapshot output The CHANGES verbiage for the "language" addition had been put under the wrong release. This moves it to be under 13.1 to 13.2 changes. ASTERISK-24553 Reported by: Matt Jordan ........ Merged revisions 429387 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	e890f9f653	ARI/AMI: Include language in standard channel snapshot output Adding information about including "language" in the standard channel snapshot output to the CHANGES file. Note the actual source changes have already been previously committed. ASTERISK-24553 Reported by: Matt Jordan ........ Merged revisions 429325 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429326 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	60ab564ad2	ari: Add support for specifying an originator channel when originating. If an originator channel is specified when originating a channel the linked ID of it will be applied to the newly originated outgoing channel. This allows an association to be made between the two so it is known that the originator has dialed the originated channel. ASTERISK-24552 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/ ........ Merged revisions 429153 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	fe6cbf455a	AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per semantic versioning, that warrants a bump in the minor version number, as it reflects a backwards compatible change. Hence, this commit. ........ Merged revisions 429091 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	fe7671fee6	Add new AMI and ARI events for connected line changes on a channel. The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 ........ Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	63cbd28999	CHANGES: Add item for new 'pjsip show identif(y\|ies) commands Tested-by: George Joseph ........ Merged revisions 428836 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428837 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago
Joshua Colp	b9aeff9580	chan_pjsip: Update CHANGES file to include 'moh_passthrough' setting git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7144c739e9	res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. This change adds a configuration option which adds a 'user=phone' parameter if the user portion of the request URI or the From URI is determined to be a number. Review: https://reviewboard.asterisk.org/r/4073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	24ded9d9eb	res_pjsip: Fix XML typo and update CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424529 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 years ago
Richard Mudgett	a47873168a	Update CHANGES for CHANNEL(onhold). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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