asterisk

Commit Graph

Author	SHA1	Message	Date
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	13 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	13 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	13 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	13 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	13 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	13 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	13 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
Richard Mudgett	73f48997f9	Add original party id and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	abf40d9b28	Add IPv6 address support to security events framework. The current Security Events Framework API only supports IPv4 when it comes to generating security events. This patch does the following: * Changes the Security Events Framework API to support IPV6 and updates the components that use this API. * Eliminates an error message that was being generated since the current implementation was treating an IPv6 socket address as if it was IPv4. * Some copyright dates were updated on files touched by this patch. (closes issue ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael L. Young Patches: security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	e4252eac10	Allow the Hangup manager action to match channels by regex * Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	655a8d4420	Introducing the log message unique call identifiers feature 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	14 years ago
Jonathan Rose	d501c2ea2d	undoing 360785 due to merging mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	bf994f0e04	Introducing the log message unique call identifiers feature 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	14 years ago
Jonathan Rose	c6979ff581	Adds F option to Bridge application Similar to dial and queue F option. (Closes issue ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1825/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	827f2eae92	Revert the pre-dial addition. The code may be just fine, but it had not received a "ship it!" on review board yet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	c65b41f57a	Add options PreDial options 'b' and 'B' to app_dial * Added 'b' and 'B' options to Dial. These options will allow you to run last-minute dialplan on the caller and callee channels while the Dial application is executing, but before the call is started. For example you can use the 'b' option to run dialplan on the callee channel to get the name of the newly created channel right away. Review: https://reviewboard.asterisk.org/r/1229/ (closes issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark Murawski, Stefan Schmidt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	c369a4416b	Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:	14 years ago
Joshua Colp	f5fda0eb74	Transition app_page to using app_confbridge internally for the conference 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	14 years ago
Richard Mudgett	a0f8821749	Add dialtone_detect option for analog incoming calls. 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	14 years ago
Sean Bright	62aae50142	Add IPv6 support to FastAGI. Review: https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon Perreault, Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	299dd5d4fc	Adds an option to sip.conf that prevents diversion headers from being added. 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	14 years ago
Tilghman Lesher	a78b0af5ea	Re-commit the verbose branch. This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	33322e38f0	res_agi: Add AGIEXITONHANGUP variable. This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is set to "yes" on a channel, AGI() will exit immediately once a channel hangup has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier and is still desired by some people. Review: https://reviewboard.asterisk.org/r/1734/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e5c51ee44c	Add auto_force_rport and auto_comedia NAT options This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3342183016	Add callbackextension matching & realtime callbackextensions This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	0adeb88318	Add CHANGES documentation for the "pri set debug" bitmask change (related to ASTERISK-17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	055a19e128	Replace res_ais with a new module, res_corosync. This patch removes res_ais and introduces a new module, res_corosync. The OpenAIS project is deprecated and is now just a wrapper around Corosync. This module provides the same functionality using the same core infrastructure, but without the use of the deprecated components. Technically res_ais could have been used with an AIS implementation other than OpenAIS, but that is the only one I know of that was ever used. Review: https://reviewboard.asterisk.org/r/1700/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	92ef8a6fe1	Address OpenSSL initialization issues when using third-party libraries. When Asterisk is used with various third-party libraries (CURL, PostgresSQL, many others) that have the ability themselves to use OpenSSL, it is possible for conflicts to arise in how the OpenSSL libraries are initialized and shutdown. This patch addresses these conflicts by 'wrapping' the important functions from the OpenSSL libraries in a new shared library that is part of Asterisk itself, and is loaded in such a way as to ensure that all calls to these functions will be dispatched through the Asterisk wrapper functions, not the native functions. This new library is optional, but enabled by default. See the CHANGES file for documentation on how to disable it. Along the way, this patch also makes a few other minor changes: * Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to more closely match what is used during run-time configuration. * Corrects some errors in the configure script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. * Adds a new variable for linker flags in the build system (DYLINK), used for producing true shared libraries (as opposed to the dynamically loadable modules that the build system produces for 'regular' Asterisk modules). * Moves the Makefile bits that handle installation and uninstallation of the main Asterisk binary into main/Makefile from the top-level Makefile. * Moves a couple of useful preprocessor macros from optional_api.h to asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	de09749470	Add an announcement option to music-on-hold - plays sound when put on hold/between songs This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a1bef6041d	Adds the ability to stop specific mixmonitors by using unique IDs set at monitor launch. MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list <channel>" which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	778fa4abaf	Various parking improvements. * Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ee4cf38a27	Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	c60d15222c	Add ABS() absolute value function to the expression parser. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	edf466012f	Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	b0243fb57c	Allow overriding of IMAP server settings on a user by user basis This patch allows the imapserver, imapport, and imapflags settings to be overridden for any voicemail user. It also documents the settings in the sample voicemail.conf file, and updates the voicemail schema to allow storage of those columns. (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	03596bcb47	chan_sip autocreatepeer=persist option for auto-created peers to survive reload This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1b0741c7db	Voicemail with the saycid option will now play a caller's name based on cid if available. In order to check the availability of the caller's name, app_voicemail will check for an audio file in <astspooldir>/recordings/callerids/ This change sets a precedent for where to put recordings of names. Currently the idea is that recordings here could also be used for applications like confbridge and meetme to find recorded names in this folder from callerid (when another recording isn't available) (closes issue ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by Russel Brown (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	9057aa20b6	Backed out core changes from r346391 During testing, it was discovered that there were a number of side effects introduced by r346391 and subsequent check-ins related to it (r346429, r346617, and r346655). This included the /main/stdtime/ test 'hanging', as well as the remote console option failing to receive the appropriate output after a period of time. I only backed out the changes to main/ and utils/, as this was adequate to reverse the behavior experienced. (issue ASTERISK-18974) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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731 Commits (95ac8f47435ebcc660a941d38ff4953c3d044e96)