summaryrefslogtreecommitdiff
path: root/CHANGES
AgeCommit message (Collapse)Author
2009-02-02Add a CLI command to log out a manager userMark Michelson
(closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02This reverts the changes I made for 11583; willSteve Murphy
reviewboard this before committing again... reopened 11583 until all Russell's issues are resolved. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02This change allows the disconnect feature (as in "one-touch" in features.c)Steve Murphy
to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30Merged revisions 172517 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29Update documentationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28Yep. Documentation is important.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27Adding AES_ENCRYPT and AES_DECRYPT dialplan functions. David Vossel
(closes issue #14301) Reported by: amorsen review: http://reviewboard.digium.com/r/128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16Fix a spelling mistake.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Related to issue #14246Olle Johansson
Update changes for SIPRemoveHeader() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13Allow specifying a port number in the user portion of a register => line in ↵Mark Michelson
sip.conf With this commit, a register => line in sip.conf may contain a port number in the "user" section of the line. Please see CHANGES and sip.conf.sample for more details regarding this. (closes issue #14198) Reported by: Nick_Lewis Patches: chan_sip.c-domainport2.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09Add a script to find out the correct settings for Asterisk behind NATMichiel van Baak
(closes issue #13065) Reported by: tzafrir Patches: sip_nat_settings uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and moi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Add the average talk time for a queueMark Michelson
This patch adds the functionality to app_queue of calculating the average amount of time that channels are bridged for a queue. The algorithm used to calculate the average is the same exponential average currently used to calculate the average holdtime. See the CHANGES file to see the methods you may use to view this information. (closes issue #13960) Reported by: coolmig Patches: app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Convert dialplan application DAHDISendCallreroutingFacility to use commas.Tilghman Lesher
(closes issue #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Fix spelling error.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Adding a new dialplan function AUDIOHOOK_INHERITMark Michelson
This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Add a new application, Originate.Russell Bryant
(closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17This patch adds a new 'ignoresdpversion' option to sip.conf. When this isMatthew Nicholson
enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Add timezone to the possible fields in a timespec.Tilghman Lesher
(closes issue #14028) Reported by: mostyn Patches: timezone-v2.patch uploaded by mostyn (license 398) (with additional code guideline fixes and a memory leak fix by me - license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Qualify trumps poke per lmadsen.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Add configuration options for finer control over how Asterisk handles having ↵Joshua Colp
to poke all peers at seemingly the same time. (closes issue #13217) Reported by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Allow disabling pattern match searches within the Realtime dialplan switch.Tilghman Lesher
(closes issue #13698) Reported by: fhackenberger Patches: 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14) Tested by: fhackenberger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12Add a new CLI command, "channel redirect", which is similar in operationRussell Bryant
to AMI Redirect. Review: http://reviewboard.digium.com/r/89/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-08Add the ability to play a courtesy tone to the transfer target in a native ↵Terry Wilson
SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-04If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) ↵Dwayne M. Hubbard
after T38 is negotiated. Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02Info on LOCAL_PEEK function.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-01Introduce CLI permissions.Eliel C. Sardanons
Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-26add support for event suppression for AMI-over-HTTPKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25Add an option, waitfordialtone, for UK analog lines which do not end a callTilghman Lesher
until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-21as suggested by jtodd, document the purposes of the CHANGES and UPGRADE filesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19Commit CHANGES change I promised when submittingMark Michelson
res_timing_timerfd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19Add info about REALTIME_FIELD and REALTIME_HASHTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12This commit does two things:Michiel van Baak
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05Add LISTFILTER dialplan function, along with supporting documentation. SeeTilghman Lesher
documentation for more information on how to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-03Adding a separation of remote authentication and our authentication.Olle Johansson
remotesecret => our password for a remote service secret => our authentication when someone calls us Secret => still has both functions if remotesecret is not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsMark Michelson
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31Failover for func_odbc, allowing an INSERT query to be performed when the ↵Tilghman Lesher
UPDATE query initially affects 0 rows. (closes issue #13083) Reported by: Corydon76 Patches: 20081031__bug13083.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30After seeing another problem in #asterisk stemming fromMark Michelson
the low default value of featuredigittimeout, I decided it was high time to change it. I have changed the default to 2000 ms based on a suggestion from Leif Madsen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30Pay attention to the searchcontexts entry in voicemail.conf (related to AST-125)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-23Thanks russellb for reminding an old man....Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-22Added debugging CLI functionsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 Give app_authenticate the ability to select a prompt other than the default. BJ Weschke
(closes issue #13734) reported and patched by: jvandal git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 The QueueEntry event now has the uniqueid of the channel included.BJ Weschke
(closes issue #13731) reported and patched by: caio1982 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Break up skinny.conf into seperate sections forMichiel van Baak
devices and lines. (closes issue #13412) Reported by: wedhorn Patches: config-restruct-v4.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Add an IAXregistry manager command. See doc/manager_1_1.txtMark Michelson
for more details of this command. (closes issue #13326) Reported by: ib2 Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-16support relative paths in musiconhold.conf, which makes moh work by default ↵Kevin P. Fleming
when Asterisk was configured using --prefix and 'make samples' is run git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14When specifying an invalid timeout to Dial, take itMark Michelson
to mean that no timeout is desired. (closes issue #13625) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-10Add keyword "same", which allows you to create multiple steps in a dialplan,Tilghman Lesher
without needing to respecify an extension pattern multiple times. (closes issue #13632) Reported by: blitzrage Patches: 20081006__bug13632.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage, Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09Add support for subscribing to a voice mailbox on a remote SIP server and ↵Joshua Colp
making the new/old message count available to local devices. (issue #AST-77) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07fix wording as pointed out by CorydonMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06This commit introduces a change to how the "joinempty"Mark Michelson
and "leavewhenempty" options are configured in queues.conf. Instead of using vague terms like "yes," "no," "loose," and "strict," we now accept a comma-separated list of values to determine when to consider a member available. Extended details can be found in the queues.conf.sample file. Note also that the above four referenced values are still accepted for backwards-compatibility, but are mapped internally to the new method of representing the option. AST-105 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-10 11:35:45 +0000