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2010-06-23Add localization support for SpanishPaul Belanger
(closes issue #17548) Reported by: cjacobsen Patches: say.conf.sample.diff uploaded by cjacobsen (license 1029) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22Add regular expression filtering for manager events.Jeff Peeler
This patch as documented in the sample config allows one to optionally apply white, black, or both types of filtering to manager events. The new 'eventfilter' option is set per user. (closes issue #14861) Reported by: fnordian Patches: eventfilter3.patch uploaded by fnordian (license 110), modified by me Review: https://reviewboard.asterisk.org/r/673/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22Merged revisions 271761 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines Allow users to specify a port for dundi peers. (closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16Merged revisions 270979 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines Fixed typo in macro-page Reported to #asterisk-dev by a student of jsmith. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Add distributed devicestate via the XMPP protocol.Tilghman Lesher
(closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Merged revisions 270442 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line Move information about zonemessages into the [zonemessages] section. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.Richard Mudgett
Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-09dial by name in chan_dahdiTzafrir Cohen
* chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Add High Resolution Times to CDRs for AsteriskBradley Latus
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change. Patch by snuffy. (closes issue #16559) Reported by: cianmaher Tested by: cianmaher, snuffy Review: https://reviewboard.asterisk.org/r/461/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Update note in sip.conf.sample.Leif Madsen
Update note in sip.conf.sample about externip and externhost with STUN. (closes issue #16323) Reported by: klaus3000 Patches: sip.conf.sample-patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Merged revisions 268320 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines Rest In Peace http://www.outandaboutnewspaper.com/article/4061 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Add ETSI Message Waiting Indication (MWI) support.Richard Mudgett
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Call Waiting support.Richard Mudgett
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Generic Advice of Charge.Richard Mudgett
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.Richard Mudgett
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.Richard Mudgett
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵Tilghman Lesher
pager messages). (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Construct socket name, according to the Postgres docs, and document as such.Tilghman Lesher
(closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24Calendaring support for Exchange Server 2007+ via EWSTerry Wilson
This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-20Add support for direct media ACLsTerry Wilson
directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address. In some networks not all phones are fully routed, i.e. not all phones can ping each other. This patch adds a way to restrict directmedia for certain peers between certain networks. (closes issue #16645) Reported by: raarts Patches: directmediapermit.patch uploaded by raarts (license 937) Tested by: raarts Review: https://reviewboard.asterisk.org/r/467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19Add ability for logger channels to include *all* levels.Kevin P. Fleming
Now that Asterisk modules can dynamically create and destroy logger levels on demand, it's useful to be able to configure a logger channel (console, file, whatever) to be able to accept log messages from *all* levels, even levels created dynamically. This patch adds support for this, by allowing the '*' level name to be used in logger.conf. Review: https://reviewboard.asterisk.org/r/663/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19fix incorrectly typed indications for [nz] stutter and dialrecallAlec L Davis
(closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17Enhancements to connected line and redirecting work.Mark Michelson
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-05Restore previous asterisk.conf syntax, where the directories aren't ↵Russell Bryant
commented out. This fixes some breakage in the test suite, that uses the contents of asterisk.conf to discover the install layout on the system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-05New static asterisk.conf.sample file.Paul Belanger
This simply moves the functionality from the Makefile (cleaning it up) into an external asterisk.conf.samples file. Also updates formatting (easier to read) and grammar changes to asterisk.conf.samples. (closes issue #17027) Reported by: pabelanger Patches: 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224) Tested by: qwell, lmadsen, pabelanger, chappell Review: https://reviewboard.asterisk.org/r/616/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵Mark Michelson
in all queues. See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-30Logic fixups for a sample FREENUM dialplan context.Tilghman Lesher
(closes issue #17263) Reported by: pprindeville Patches: freenum-dialplan.patch#3 uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-29Pattern match fail.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-27Merged revisions 259270 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-20Add 'soft hangup' alias per Steve Johnson on asterisk-users.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-20Add example dialplan for dialing ISN numbers (http://www.freenum.org).Leif Madsen
Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-18Removing unused configuration parametersTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-14Yet another issue where the conversion of the application delimiter to comma ↵Tilghman Lesher
caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-13Merged revisions 257070 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-13Update sample dialstrings in sip.conf.sample file.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09Merge Call completion support into trunk.Mark Michelson
From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-01Removed documentation of the non existent 'both' option to 'faxdetect' in ↵Matthew Nicholson
sip.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Add documentation clarifying when 't' and 'T' can be used.Leif Madsen
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Replace some documentation from 1.6.x back into trunk.Leif Madsen
This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Update confusing documentation for tlsbindaddr.Leif Madsen
Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capableKevin P. Fleming
application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.Jeff Peeler
Previously only configurable globally. A unit test has also been written to provide protection against parse failures for supported mailbox options. (closes issue #16864) Reported by: kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/555/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-22Accomodate equal signs in DSNs and add documentation, based upon ↵Tilghman Lesher
mmichelson's feedback. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-17Merged revisions 253018 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-16Merged revisions 252761 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-15Merged revisions 252533 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-14generate roundtrip delay requests and responsesAlexandr Anikin
added response to roundtrip delay requests from opposite side added roundtrip delay request sending to opposite side after answer, added options for sending request (interval between request and count of unreplied requests before forced call hangup) (closes issue #16976) Reported by: vmikhelson Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454) Tested by: vmikhelson, may213 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-12Only change the RTP ssrc when we see that it has changedTerry Wilson
This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-03Add new config option to control AMI alarm event reporting in chan_dahdi.Jeff Peeler
New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Merge missed files from res_fax/res_fax_spandsp merge.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-04 23:25:14 +0000