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2012-09-05LDAP Realtime Peers Cannot RegisterDarren Sessions
Prior to 1.8, it was not necessary for an explicit "type" to be set for an asterisk LDAP realtime peer. Now the routine find_peer actually checks the type field during registration and fails to find the peer if it is not set. The attached patch makes the realtime type equal whatever type is being searched for if the type is 0 upon return from routine build_peer. (closes issue ASTERISK-17222) Reported by: John Covert Patch by: David Vossel Tested by: Darren Sessions Review: https://reviewboard.asterisk.org/r/2095/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-27Fix misleading documentation in agents.conf.sample regarding ackcall usage.Mark Michelson
The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371789 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371790 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-27Fix incorrectly documented option in queues.confMark Michelson
sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371748 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371750 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07Add named callgroups/pickupgroupsMatthew Jordan
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Add headers from SIPAddHeader to outbound REFER requests.Mark Michelson
This is a patch from kkm from review board. This is useful for adding headers to REFER requests that emanate from a Transfer() dialplan application call. This also fixes some uses of the Referred-by header, removing an extra set of angle brackets. I've modified the reporter's original patch to not require any additions to the sip_refer header and to just remove the referred_by_name from sip_refer since it is no longer needed or used. (closes Issue ASTERISK-17639) reported by Kirill Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff uploaded by Kirill Katsnelson (license #5845) Review: https://reviewboard.asterisk.org/r/1159 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Add "setvar" option to manager.conf.Mark Michelson
With this option set, channel variables can be set on every manager originate. The Variable header can still be used to set additional channel variables for individual calls if desired. This work was completed by Olle Johansson on review board. I have applied the review feedback and am committing it in order to get this into trunk before Asterisk 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-23Add separate configuration options for subscription and registration ↵Mark Michelson
minexpiry and maxexpiry. This offers more fine-grained control over how long subscriptions last without negatively affecting the expiration range for registrations. Uploaded by: Guenther Kelleter(license #6372) Review: https://reviewboard.asterisk.org/r/2051 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-22Prevent multiple local candidates from being added with the same information ↵Joshua Colp
and add support for disabling ICE on a per-peer basis. (closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19named_acl: Remove systemname option from acl.conf, use asterisk.conf valueJonathan Rose
Review: https://reviewboard.asterisk.org/r/2057/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Added option 'interdigit_timer' to unistim.conf to make able controll ↵Igor Goncharovskiy
hardcoded dial timeout constant. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add support for SIP over WebSocket.Joshua Colp
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Handle deprecated (aliased) option names with the config options apiTerry Wilson
Add a simple way to register "deprecated" option names that alias to a different "current" name. Review: https://reviewboard.asterisk.org/r/2026/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Named ACLs: Introduces a system for creating and sharing ACLsJonathan Rose
This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Document that multiple endpoints using the same connection is not supported.Joshua Colp
(closes issue ASTERISK-20104) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add Digium phones context to sip_notify sample config.Jason Parker
This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-07Add a new unified Jingle, Google Jingle, and Google Talk channel driver ↵Joshua Colp
written from scratch called chan_motif. This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either. These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold, unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications. The original Google Talk protocol is also supported for Google Voice interoperability. You may ask yourself though where the name motif comes from... and I would say to you... music! motif: a perceivable or salient recurring fragment or succession of notes Sorta like a jingle! Review: https://reviewboard.asterisk.org/r/1917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-04Added direct media support to ooh323 channel driverAlexandr Anikin
options are documented in config sample sample config rename to proper name - ooh323.conf To change media address ooh323 send empty TCS if there was completed TCS exchange or send facility forwardedelements with new fast start proposal if not. Then close transmit logical channels and renew TCS exchange. If new fast start proposal is received then ooh323 stack call back channel driver routine to change rtp address in the rtp instance. If empty TCS is received then close transmit logical channels and renew TCS exchange Review: https://reviewboard.asterisk.org/r/1607/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-02Add a cleaned up drop-in replacement for res_jabber called res_xmpp. This ↵Joshua Colp
provides the same externally facing functionality but is implemented differently internally. This is currently not built by default but this will be changed once chan_jingle2 (insert actual name in your head when reading this after it has been merged) is in the tree. Review: https://reviewboard.asterisk.org/r/1983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Hangup handlers - Dialplan subroutines that run when the channel hangs up.Richard Mudgett
Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-28Add the ability to set flags via the config options apiTerry Wilson
Allows the setting of flags via the config options api. For example, code like this: #define OPT1 1 << 0 #define OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int flags; }; and a config like this: [blah] opt1=yes opt2=no opt3=yes Review: https://reviewboard.asterisk.org/r/2004/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04Merge changes dealing with support for Digium phones.Mark Michelson
Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-01Add missing config for config API testTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-01Add new config-parsing frameworkTerry Wilson
This framework adds a way to register the various options in a config file with Asterisk and to handle loading and reloading of that config in a consistent and atomic manner. Review: https://reviewboard.asterisk.org/r/1873/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-01Help mitigate potential reinvite glare scenarios.Mark Michelson
When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. Review: https://reviewboard.asterisk.org/r/1954 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18app_queue: Per Member ringinuse option and deprecation of ignorebusyJonathan Rose
Adds a number of methods for controlling the setting of 'ringinuse' which is basically the same concept as the old ignorebusy setting, only now the per member setting always controls whether or not the member is actually ringed while in use. A CLI command and a manager action have been added to change a given queue member's ringinuse option while Asterisk is running and the an argument has been added for adding members with deliberately set ringinuse in queues.conf Some effort has been made to ensure compatability with dialplans and databases still referring to 'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1919/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Improve FollowMe accept/decline DTMF string matching.Richard Mudgett
If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Keep answered FollowMe calls until call accepted or last step times out.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28Add support for lightweight NAT keepalive.Joshua Colp
If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Make it possible to change the minimum DTMF duration in asterisk.confOlle Johansson
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19Update membermacro and membergosub documentation in queues.conf.sample.Richard Mudgett
........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362678 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-18Add ability to ignore layer 1 alarms for BRI PTMP lines.Richard Mudgett
Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12Add option to invoke the extensions.conf stdexten using the legacy macro method.Richard Mudgett
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-11Change default value of 'ignorebusy' on Queue members so that behavior is ↵Jonathan Rose
more like 1.8 Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged revisions 361907 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06Remove a few more files related to chan_usbradio and app_rpt.Russell Bryant
........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361381 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29undoing 360785 due to merging mistakeJonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29Introducing the log message unique call identifiers featureJonathan Rose
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Add ability for chan_dahdi ISDN to block connected line updates per span.Richard Mudgett
Added new chan_dahdi.conf colp_send option parameter to block connected line updates per span. (closes issue ASTERISK-17025) Reported by: Michael Smith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-12Massive changes in chan_unistim channel driver. Include many fixes in ↵Igor Goncharovskiy
channel driver operation and add additional functionality: * Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls. * Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone * Other described in CHANGES file Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E. (closes issue ASTERISK-16890) Review: https://reviewboard.asterisk.org/r/1243/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-10Transition app_page to using app_confbridge internally for the conference ↵Joshua Colp
bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles. Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-07Handle numeric columns for eventtype properly in cel_odbcTerry Wilson
Patch also implements correct handling of datetime2 and datetimeoffset new datatypes in SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548) Review: https://reviewboard.asterisk.org/r/1160/ Review: https://reviewboard.asterisk.org/r/1804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-07Set snarkiness = 0 in cdr_adaptive_odbc.conf.sampleTerry Wilson
........ Merged revisions 358438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358441 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-06Add dialtone_detect option for analog incoming calls.Richard Mudgett
For analog lines, enables Asterisk to use dialtone detection per channel if an incoming call was hung up before it was answered. If dialtone is detected, the call is hung up. no: Disabled. (Default) yes: Look for dialtone for 10000 ms after answer. <number>: Look for dialtone for the specified number of ms after answer. always: Look for dialtone for the entire call. Dialtone may return if the far end hangs up first. dialtone_detect=yes dialtone_detect=5000 dialtone_detect=always (closes issue ASTERISK-19316) Reported by: Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy Pepper Review: https://reviewboard.asterisk.org/r/1737/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-05Tab to spaces and text change.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02Beef up the IAX2 sample configuration a bit and fix some formatting issues.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Adding transport=udp to sample sip.conf - Also changes version of Asterisk ↵Jonathan Rose
1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Deprecated macro usage for connected line, redirecting, and CCSSKinsey Moore
This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Adds an option to sip.conf that prevents diversion headers from being added.Jonathan Rose
send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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