asterisk

Commit Graph

Author	SHA1	Message	Date
Paul Belanger	67d2ce9e73	authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	be5c67401d	Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 \| mjordan \| 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) \| 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbfbbbeb32	Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 \| rmudgett \| 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) \| 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d367c2631c	Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 \| rmudgett \| 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) \| 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b83a26eefa	In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 \| mnicholson \| 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) \| 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6f96c25efa	Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 \| lathama \| 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) \| 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6e937b1b63	Revert merge of r306999, due to merge conflict. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	55e0af4d27	The default conf-usermenu says that '8' can be used to leave the conference, so put that in the sample user menu. '5' is supposed to extend the conference, but there doesn't appear to be a concept of that in the menu actions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1ddcb689d6	Merged revisions 329709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329709 \| jrose \| 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) \| 8 lines Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf (closes issue ASTERISK-16263) Reported by: richardf Patches: nz-indications.patch uploaded by richardf (License #6015) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	361d40e7fb	Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 \| jrose \| 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) \| 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	46dd023878	Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 \| rmudgett \| 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) \| 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a97340b5ea	Merged revisions 328014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328014 \| rmudgett \| 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) \| 1 line Add ATXFER_NULL_TECH note in features.conf.sample. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	fe084047ee	Full T.38 handshaking and fax detection Add full t.38 handshaking for OOH323 that are required for newest T.38 gateway codes. Add fax detection (cng tone, t38) and dialplan redirection to fax ext on fax event detected. Add OOH323() function to set/get t38support and faxdetect parameters. (closes issue ASTERISK-17754) Reported by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot (license 52) issue19183-final.patch uploaded by may213 (license 454) Tested by: may213, irroot Review: https://reviewboard.asterisk.org/r/1174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	17860b70e4	Updates confbridge.conf video documentation and adds dtmf action for releasing video src. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	39a7152df3	Merged revisions 325935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 \| rmudgett \| 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) \| 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	1339a0a535	Video support for ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	f99a06d030	Commit "distrotech" app_queue changes to Trunk	14 years ago
Leif Madsen	92dcabe726	Merged revisions 324241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324241 \| lmadsen \| 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) \| 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	0bd877621e	Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5cb2775480	Merged revisions 322189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 \| pabelanger \| 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) \| 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a0468ca7fa	Merged revisions 321685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 \| lmadsen \| 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) \| 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f90bc95f0d	Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 \| jrose \| 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) \| 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5257a915a8	Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6eb9d7e1b5	Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 \| jrose \| 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) \| 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	07ba8b1474	Updated the sample pbx_lua config file to reflect autoservice changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317818 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
Leif Madsen	c85a903198	Merged revisions 317058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317058 \| lmadsen \| 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) \| 7 lines Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 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	37274c73ee	Problems with ISDN MWI to phones. The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ae2926b5d0	Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => 823000,hint,ccss:sip/3000 You would then subscribe a BLF button to 823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	a77fd545ab	Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 \| mnicholson \| 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) \| 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 \| mnicholson \| 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) \| 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 \| mnicholson \| 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) \| 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	2176df5d83	Merged revisions 311930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311930 \| tilghman \| 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) \| 6 lines Incorrect default example; the field is actually internally named "clid", not "callerid". (closes issue #19040) Reported by: wcselby Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	08828045b1	Merged revisions 311050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311050 \| alecdavis \| 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) \| 24 lines Merged revisions 311049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 \| alecdavis \| 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) \| 17 lines Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 \| alecdavis \| 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) \| 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	0a96892b04	Merged revisions 309765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309765 \| mmichelson \| 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) \| 3 lines Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	01a453351d	Add setvar option to calendaring Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b20fecdbbb	Add support for defining hints from pbx_lua (closes issue #16024) Reported by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5deb544d06	Merged revisions 308679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 \| twilson \| 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) \| 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 \| twilson \| 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) \| 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	4cba13eb60	Merged revisions 307467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307467 \| mmichelson \| 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) \| 5 lines Fix a gaffe in the CCSS sample configuration. Discovered by Philippe Lindheimer and pointed out on #asterisk-dev ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	a350924700	Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8b584000a9	Define the MCID acronym in chan_dahdi.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a8aeb04a9f	Add ISDN display ie text handling options to chan_dahdi.conf. The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	652fb64a01	res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support (issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	93bade5639	Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	9f1a17f137	Replacing doc/* with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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1481 Commits (04224c20362f94f99886d6b51adb7b90db62855f)