asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	9d79ccd3db	Add UPGRADE-11.txt file; update UPGRADE.txt to reflect Asterisk 12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	014e8a0a80	Add notes to UPGRADE.txt about addition of msg_id to VoiceMails. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	941941ce98	Update UPGRADE.txt with notes about ICE support and res_xmpp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	cb9756daa2	Add hangupcause translation support The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 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
Kinsey Moore	afa03bd310	Parse ANI2 information from SIP From header parameters ANI2 information is now parsed out of SIP From headers when present in the oli, isup-oli, and ss7-oli parameters and is available via the CALLERID(ani2) dialplan function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon Review: https://reviewboard.asterisk.org/r/1947/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	cc69a0deaf	Document BLINDTRANSFER behavior change. (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b5a6de76fc	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 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	13 years ago
Richard Mudgett	d71d8ed995	Keep answered FollowMe calls until call accepted or last step times out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 years ago
Richard Mudgett	a35c7ba8e7	Add option to invoke the extensions.conf stdexten using the legacy macro method. 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	13 years ago
Sean Bright	3a231e090f	chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus AMI Events The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. Post was removed and the AMI version has been updated to 1.3. ........ Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	c369a4416b	Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:	13 years ago
Kinsey Moore	1fac2fba4b	Deprecated macro usage for connected line, redirecting, and CCSS 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	13 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
Joshua Colp	afdd96712c	Make the 'c' option to MeetMe work even if the 'q' option is used. (closes issue ASTERISK-17053) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354084 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
Kinsey Moore	71a8457d53	Support schema selection in cdr_adaptive_odbc Asterisk now supports using ODBC with databases where a single schema must be selected. Previously, INSERTs would fail because they did not take into account extra fields cause by having multiple schemas. This also corrects some SQL resource leaks. (closes issue ASTERISK-17106) Patch-by: Alexander Frolkin Patch-by: Tilgnman Lesher git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c6fd4f5d74	SIP session timeout AMI event Add an AMI event in the Call category that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration. Event description: Event: SessionTimeout Source: source Channel: channel-name Uniqueid: channel-unique-id `source` can be either RTPTimeout or SIPSessionTimer (closes issue ASTERISK-16467) Patch-by: Kirill Katsnelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 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
Richard Mudgett	d6b359ff0b	Make pbx_config.c use Gosub instead of Macro call for stdexten. Users created by users.conf with hasvoicemail=yes have been documented as using a Gosub to stdexten since v1.6.0. However, the code still generates dialplan to access stdexten as a Macro as documented in v1.4; which does not work with the newer extensions.conf.sample file. * Make generated dialplan access the stdexten dialplan with the documented Gosub instead of the older Macro style. (closes issue ASTERISK-18809) Reported by: Jay Allen Patches: gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349782 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
Walter Doekes	fd64bb66f9	Add VM_INFO() dialplan function to gather information about a mailbox. Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	77b670c4ab	Allow each logging destination and console to have its own notion of the verbosity level. Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	00a522c000	Correct the default udptl port range. The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 ........ Merged revisions 343580 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	15fd1e375c	Return error when no rows are deleted for AMI DBDelTree (closes issue AST-654) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6708ee76a0	Merged revisions 340219-340220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340219 \| twilson \| 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) \| 8 lines Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ ........ r340220 \| twilson \| 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) \| 2 lines Add a missing file for the astdb2bdb conversion utility ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2e2381341e	Clean up dsp.conf parsing Review: https://reviewboard.asterisk.org/r/1434/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2d18de5f8f	Clean up cdr.conf parsing for [csv] section Review: https://reviewboard.asterisk.org/r/1427/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	61b369ac76	Clean up dnsmgr.conf parsing Review: https://reviewboard.asterisk.org/r/1432/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	749ef800aa	Be more specific on which section has changed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	b52b026a35	Iterate though cdr.conf setting Review: https://reviewboard.asterisk.org/r/1426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	052ece39ee	Merged revisions 332029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332029 \| mnicholson \| 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) \| 2 lines Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c3bd5892a6	Allow ENUM query functions to report lookup errors The ENUM dialplan functions do not report DNS query errors properly. It is useful to differentiate between failed query (e.g. non-existent domain) vs. no data records of the appropriate type. This is required to make overlapped dialing work. (closes issue ASTERISK-13769) Review: https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	16acfefa74	Merged revisions 331097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 \| twilson \| 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) \| 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	2c198555fd	Fix UPGRADE.txt files for Asterisk 10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	1f65d55fb0	Merged revisions 328448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328448 \| lmadsen \| 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) \| 2 lines Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	c98447f82c	Add UPGRADE-1.10.txt file from UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	13f92d2b82	Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	efd040cd11	Replace Berkeley DB with SQLite 3 There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	f99a06d030	Commit "distrotech" app_queue changes to Trunk	14 years ago
Richard Mudgett	cdee44e992	Merged revisions 321337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 \| rmudgett \| 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) \| 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d211be98ed	Add note about PrivacyManager to UPGRADE.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	7a1204d129	Default to starting an autoservice in pbx_lua. The autoservice is automatically stopped when applications are executed, so this shouldn't cause any problems. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	d5e9ce9ab1	Make pbx_lua handle managing the autoservice better. Make autoservice_start() and autoservice_stop() return nothing. Also check if the autoservice flag is set before starting or stopping the autoservice and stop and start the autoservice when returning control to and getting control from the pbx engine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	6c38322870	Added note about changes in pbx_lua's behavior when applications do dialplan jumps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2dfb427540	Add CEL extra field to cel_pgsql. (closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	90177fe708	Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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241 Commits (837e00a5cc90699a62e93f15dce463e27ece250e)