asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	ff09fa5ac5	chan_sip: Document a change to user-field encoding introduced with r303509 The change in question was added to improve compliance with RFC3261, but at the time of commit, it wasn't adequately documented in the UPGRADE notes. (closes issue ASTERISK-20561) Reported by: Deniz Review: https://reviewboard.asterisk.org/r/2177/ ........ Merged revisions 375846 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375847 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	da85f8489f	Make evaluation of channel variables consistently case-sensitive. Due to inconsistencies in how variable names were evaluated, the decision was made to make all evaluations case-sensitive. See the UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for more details. (closes issue ASTERISK-20163) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2160 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1eb14dbff8	Ensure that CDRs for a caller in a Queue that is not answered is NO ANSWER. When a caller enters a queue and no queue member answers the call, the current behaviour can be a little odd depending on the paused status of the queue members. If any queue member is paused, but not all, the CDR disposition will be BUSY. If all queue members are paused, then the CDR disposition is based instead on the disposition of the call prior to entering the Queue. This patch modifies the behaviour in the following ways: * If no queue members are paused, the CDR disposition is whatever the disposition was prior to going into Queue. If the call was answered this will be ANSWERED; otherwise, it is NO ANSWER. * If some queue members are pused, the CDR result is NO ANSWER. (This is a change in behaviour, as the result would previously have been BUSY) * If all queue members are paused, the CDR result is whatever the result was prior to going into Queue. This is the same as the behaviour prior to this patch. * If the caller hangs up, times out, or presses '*' with the 'h' option, the CDR disposition is again not set and is dependent on whether or not the caller was Answered prior to entering Queue. This patch was based on one provided by Thomas Arimont, but has been modified to accomodate findings by the reviewers. Review: https://reviewboard.asterisk.org/r/2064/ (closes issue AST-906) Reported by: Thomas Arimont (closes issue ASTERISK-17776) Reported by: Attila Megyeri git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	987a5d58c3	app_queue: add upgrade notes for 375216 Adds UPGRADE notes describing behavioral changes to rrmemory strategy caused by 375216 (issue AST-989) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b5138fccf4	Add pause one second W dial modifier. * The following dialplan applications now recognize 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. * Created dahdi_dial_str() in chan_dahdi that eliminated a lot of duplicated dialing code and diagnostic messages for the channel driver. (closes issue ASTERISK-20039) Reported by: Jeremiah Gowdy Patches: jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy Expanded patch to add support in chan_dahdi. Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 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	4272ec14c5	app_queue: PAUSEALL/UNPAUSEALL logged only if interface is a queue member Adding UPGRADE.txt entry for r372148 (issue AST-946) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	6c07c904aa	chan_sip: Change manager event to confirm SIPqualifypeer into an ack Matt Jordan informed me that it was more appropriate to use an astman_send_ack here instead of making an event response. I've also used this opportunity to update UPGRADE.txt to mention this change in behavior. (issue AST-969) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
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	13 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

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249 Commits (f28061145f16d834d0ecee4d8c5bb605d957d7f8)