763 Commits (099cd07887a1af2ead77a60f744658b7077ed58f)

Author	SHA1	Message	Date
Jonathan Rose	ae655031b9	Features: BRIDGE_FEATURES variable automixmonitor support and use proper party ... BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it does. In addition, the BRIDGE_FEATURES variable would not apply features to the proper party based on whether the feature option letter was in caps or in lowercase (both ways would apply it to the caller). Now uppercase applies to the caller while lowercase applies to the callee (like with the dial option) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	e62bab8131	chan_sip: Add SubscribeContext field to SIPshowpeer AMI response ... The new field is will show up within the response if the requested peer has a subscribe context set. (closes issue ASTERISK-20626) Reported by: Jaco Kroon Patches: asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671) -with modifications by jrose to conform to style guidelines Review: https://reviewboard.asterisk.org/r/2195/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	b2f9542f61	manager: Change display of 'manager show commands' and 'manager show command' ... manager show commands now shows the full name of the command being displayed regardless of size. The privilege column has also been removed from this display. It will also now use the full length of the terminal if curses is available. Manager show command will now always display the privilege of the manager command within the CLI. (closes ASTERISK-20396) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/2143/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	90f8c90b10	dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END ... Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374481 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374485 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	4af961a03a	dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values ... Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	acb3a5f76f	Add Duration header for PlayDTMF AMI Action ... This patch adds an optional header to the PlayDTMF AMI action, Duration. It allows the duration of the DTMF digit to be played on the channel to be specified in milliseconds. (closes issue ASTERISK-18172) Reported by: Renato dos Santos patches: send-dtmf.patch uploaded by Renato dos Santos (license #6267) Modified slightly for this commit for Asterisk 12. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	5bde2dbc34	Add VoicemailRefresh AMI Action ... Currently, if there are modifications to mailboxes that Asterisk is not aware of, the user needs to add "pollmailboxes" to their mailbox configuration, which repeatedly polls the subscribed mailboxes for changes. This results in a lot of extra work for the CPU. This patch introduces the AMI command VoicemailRefresh which permits external applications to trigger the refresh themselves. The refresh can apply to a specified mailbox only, an entire context, or all configured mailboxes. Even a refresh performed on every mailbox would not consume as much CPU as the pollmailboxes option, given that pollmailboxes runs continuously and this only runs on demand. (closes issue ASTERISK-17206) (closes issue ASTERISK-19908) Reported-by: Jeff Hutchins Reported-by: Tilghman Lesher Patch-by: Tilghman Lesher git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fdfb3ae5fa	Allow for redirecting reasons to be set to arbitrary strings. ... This allows for the REDIRECTING dialplan function to be used to set the reason to any string. The SIP channel driver has been modified to set the redirecting reason string to the value received in a Diversion header. In addition, SIP 480 response reason text will set the redirecting reason as well. (closes issue AST-942) reported by Malcolm Davenport (closes issue AST-943) reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/2101 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ca0e96ae19	Add queue monitoring hints ... This patch adds support for hints on a queue. Hints can be added using the nomenclature 'Queue:name', where name is the name of the queue being monitored. This nifty feature was done by Alec Davis. Review: https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis Tested by: alecdavis patches: review1619.diff2 by alecdavis (license 585) ........ Merged revisions 373235 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	67ca3b9126	app_queue: Support an 'agent available' hint ... Sets INUSE when no free agents, NOT_INUSE when an agent is free. modifes handle_statechange() scan members loop to scan for a free agent and updates the Queue:queuename_avial devstate. Previously exited early if the member was found in the queue. Now Exits later when both a member was found, and a free agent was found. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2121/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a05f001ba2	chan_sip: Fix CHANGES and UPGRADE.txt for r372808 ... (issue AST-969) Reported by John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	b02c65752c	app_queue: Only log PAUSEALL/UNPAUSEALL when 1+ memebers changed. ... Prior to this patch, if pause or unpause was issued on an interface without specifying a specific queue, a PAUSEALL or UNPAUSEALL event would be logged in the queue log even if that interface wasn't a member of any queues. This patch changes it so that these events are only logged when at least one member of any queue exists for that interface. (closes issue AST-946) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2079/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e2cd045c7e	Update CHANGES for private party ID. ... ........ Merged revisions 371146 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1067294065	DUNDi: Add CLI commands DUNDi show cache and DUNDi show hints ... (closes issue ASTERISK-18390) Reported by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290) ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	b03e7cc4c7	Move a SIP change up to the other SIP changes in the CHANGES file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eb9e645a27	Allow support for early media on AMI originates and call files. ... This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	ee849b461f	Add AMI_CLIENT dialplan function ... Implementation of a dialplan function for checking manager accounts. Right now it only returns the number of logged in sessions for a manager account, but other attributes can be added later. Patch by: Olle Johansson Review: https://reviewboard.asterisk.org/r/421/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c4578f4ad	Add named callgroups/pickupgroups ... 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	13 years ago
Mark Michelson	4377d511ae	Add headers from SIPAddHeader to outbound REFER requests. ... 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	13 years ago
Mark Michelson	58f281a670	Add "setvar" option to manager.conf. ... 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	13 years ago
Mark Michelson	6c23a60f80	Add "dialplan remove context" and modify "dialplan add include" ... From corruptor's review board posting: "I've noticed that we can remove particular extension from context with dialplan remove extension command but in order to remove all extensions in the context we should delete them on by one. I've created dialplan remove context command which uses ast_context_destroy to destroy the whole context with all extensions. I've created to functions for in pbx_config.c: handle_cli_dialplan_remove_context which actually removes context and complete_dialplan_remove_context which completes input. They are based on other similar functions and pretty trivial but I can be mistaken somewhere. "I've also modified dialplan add include <context2> into <context1>. I've made it similar dialplan add extension ... command. It creates <context1> if it doesn't exist and I've also modified complete_dialplan_add_include and removed check for existance of <context2> because we can include non-existent context into another one. (I usually include empty (non-existent) contexts in advance). Should we raise warning in this case as it's raised while reading extensions.conf? "I use those functions with AMI. I think manager commands should be created in addition to those CLI commands." I've addressed the latest comments on review board and have made some other coding guidelines-related cleanup. I also have modified the CHANGES file to mention these new commands. (closes issue ASTERISK-19292) reported by Andrey Solovyev Patches: dialplan_add_include.patch uploaded by Andrey Solovyev (license #5214) dialplan_remove_context.patch uploaded by Andrey Solovyev (license #5214) Review: https://reviewboard.asterisk.org/r/2042 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	9850a075b7	Allow specifying a port number for the MySQL server. ... This patch allows you to specify a port number for the MySQL server. It's useful if a MySQL server is running on a non-standard port. Even though this module is deprecated in favor of func_odbc, someone asked for this feature and it seems pretty harmless to add. It has been tested using a number of combinations of with/without a port number specified in the dialplan and changing the port number for mysqld. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	3da07b3ec0	chan_sip: Add SIPpeerstatus command to AMI ... This patch was submitted by mnicholson a while back. It adds a new AMI action which allows users to request SIP peer status on demand similar to existing PeerStatus events and to the output you would see from CLI with sip show peer Review: https://reviewboard.asterisk.org/r/1098/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	7d4ccea736	Enable usage of system-provided NetBSD editline library if available. ... This patch changes the Asterisk configure script and build system to detect the presence of the NetBSD editline library (libedit) on the system. If it is found, it will be used in preference to the version included in the Asterisk source tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie Review: https://reviewboard.asterisk.org/r/1528/ Patches: 0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	b74acabdc7	Update CHANGES for list/negation ACL feature. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	b5193428a7	Enable usage of system-provided iLBC library. ... The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a28e6fc7bd	Add separate configuration options for subscription and registration 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	13 years ago
Matthew Jordan	92abf49458	Update CHANGES for Asterisk 11 ... This updates the CHANGES file with things that were committed for Asterisk 11, but were not noted in that file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	499a445af2	Update CHANGES about adding the AccountCode header to the AMI Hangup event. ... (issue ASTERISK-19963) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	9278b5e51e	Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant. ...	13 years ago
Kinsey Moore	25e721ee9b	Add comments about the BUILD_NATIVE change ... This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e938737570	Add support for SIP over WebSocket. ... 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	13 years ago
Igor Goncharovskiy	95ac8f4743	Add French translation for chan_unistim phones on-screen menus. ...	13 years ago
Jonathan Rose	10afdf3a2a	Named ACLs: Introduces a system for creating and sharing ACLs ... 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	14 years ago
Joshua Colp	a3fa37b8cf	Add a new unified Jingle, Google Jingle, and Google Talk channel driver 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	14 years ago
Matthew Jordan	3044aa3e38	Add 'stun show status' command ... This patch adds a new CLI command, 'stun show status'. This command will show a table describing all known STUN servers and statuses. (closes issue ASTERISK-18046) Reported by: Jeremy Kister Tested by: Jeremy Kister patches: (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy Kister) Review: https://reviewboard.asterisk.org/r/2001 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	fa10f3f8a8	Added direct media support to ooh323 channel driver ... 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	14 years ago
Richard Mudgett	ac35b92b62	Hangup handlers - Dialplan subroutines that run when the channel hangs up. ... 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	14 years ago
Matthew Jordan	82a7409c15	Add AMI event documentation ... This patch adds the core changes necessary to support AMI event documentation in the source files of Asterisk, and adds documentation to those AMI events defined in the core application modules. Event documentation is built from the source by two new python scripts, located in build_tools: get_documentation.py and post_process_documentation.py. The get_documentation.py script mirrors the actions of the existing AWK get_documentation scripts, except that it will scan the entirety of a source file for Asterisk documentation. Upon encountering it, if the documentation happens to be an AMI event, it will attempt to extract information about the event directly from the manager event macro calls that raise the event. The post_process_documentation.py script combines manager event instances that are the same event but documented in multiple source files. It generates the final core-[lang].xml file. As this process can take longer to complete than a typical 'make all', it is only performed if a new make target, 'full', is chosen. Review: https://reviewboard.asterisk.org/r/1967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	babc0983e8	Add IPv6 Support To Manager ... This patch adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported by: Michael L. Young Tested by: Michael L. Young Patches: ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1968/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	37677a8cc2	Merge 'core' and 'core changes' sections in CHANGES file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ec3b8a1f27	app_queue: Per Member ringinuse option and deprecation of ignorebusy ... 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	14 years ago
Jonathan Rose	cd37bec058	logger: Adds additional support for call id logging and chan_sip specific stuff ... This patch improves the handling of call id logging significantly with regard to transfers and adding APIs to better handle specific aspects of logging. Also, changes have been made to chan_sip in order to better handle the creation of callids and to enable the monitor thread to bind itself to a particular call id when a dialog is determined to be related to a callid. It then unbinds itself before returning to normal monitoring. review: https://reviewboard.asterisk.org/r/1886/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1ae31fd2a9	Add predial support to FollowMe. ... Like the new predial feature for Dial. This adds the same b/B options to FollowMe. Review: https://reviewboard.asterisk.org/r/1910/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	69e2a2b265	Update documentation references in CHANGES to reflect the correct pages on the wiki. ... The current CHANGES file refers to doc/ in many places and those files no longer exist. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	474612d7f7	Add IPv6 support to ExternalIVR. ... Review: https://reviewboard.asterisk.org/r/1896/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	ae1502be33	Add support for lightweight NAT keepalive. ... 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	14 years ago
Russell Bryant	386c2c6edf	features: Add FEATURE() and FEATUREMAP() functions. ... Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the [general] section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. See the built-in documentation for details. Review: https://reviewboard.asterisk.org/r/1871/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e8a6e0ef0e	PreDial - Ability to run dialplan on callee and caller channels before Dial. ... Thanks to Mark Murawski for the initial patch and feature definition. (closes issue ASTERISK-19548) Reported by: Mark Murawski Review: https://reviewboard.asterisk.org/r/1878/ Review: https://reviewboard.asterisk.org/r/1229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7aa0c3c64b	Make it possible to change the minimum DTMF duration in asterisk.conf ... 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	14 years ago