asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	ce2f670228	Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level. (Closes issue #11015) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	fd4f9d55e8	Remove second prefix line. Only need it documented once in the same file. (closes issue #11472) Reported by: eserra Patches: http.conf.sample.diff uploaded by eserra (license 45) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0cc002a48a	Rename "username" to "defaultuser" to match with "defaultip". "Username" still works, but is deprecated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f15be28fb0	Add support for monitoring MWI on FXO lines. This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	18259c2318	Updating sample queues.conf file to show how multiple periodic announcements may be specified since this was not documented previously (closes issue #11432, reported and patched by Laureano) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	6b08c442c7	Adding support for the "automixmonitor" dial and queue options. This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	57c2bcca86	Merged revisions 90098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 \| kpfleming \| 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) \| 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	a42259c3ff	Adding support for realtime music on hold. The following are the main points: 1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	df1689e927	Merged revisions 89634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 \| russell \| 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) \| 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	b1c0c67e76	Merged revisions 89624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 \| oej \| 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) \| 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	4d8932a6dc	Merged revisions 89622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 \| murf \| 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) \| 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	11df6a9119	Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated. Both still works in this version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	2ec4b57622	Thanks to pnlarsson for noting the spelling error in the cli commands. Also, added some verbage about the new algorithm to CHANGES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f1de129e5f	Merged revisions 89559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89559 \| tilghman \| 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) \| 14 lines We previously attempted to use the ESCAPE clause to set the escape delimiter to a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	07cb09ad86	- Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	a63f6be669	closes issue #11363 ; where the pattern _20x. buried in an included context, didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f0780d2b47	Merged revisions 89527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89527 \| russell \| 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) \| 3 lines mvanbaak pointed out a spelling error in this sample configuration file. While I was at it, I went ahead and tweaked it a little bit more. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	f5e5a443cf	Changed occurrences of "busy-level" to "busylevel" in sip.conf.sample in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	eab6b00904	Add support for application/dtmf SIP INFO dtmf handling. Yep, another way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Christian Richter	2a0b16b663	Merged revisions 89173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 \| crichter \| 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) \| 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Christian Richter	c9b8afb447	Merged revisions 89169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 \| crichter \| 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) \| 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	a442780a75	Add usbradio.conf.sample from branches/1.4/configs - r84162. It was mistakenly deleted in 1.4 without ever being merged to trunk. Reported by eliel on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b436362b19	Fix a few potential deadlocks in cdr_sqlite3_custom. (also rename sample config to .sample) Closes issue #11208, patch by Laureano. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	e03cb6a721	Merged revisions 89115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11195) ........ r89115 \| qwell \| 2007-11-08 12:45:15 -0600 (Thu, 08 Nov 2007) \| 4 lines Avoid warnings on load when using sample configuration files. Issue 11195, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6a9fbeaf68	Merged revisions 89079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89079 \| tilghman \| 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) \| 5 lines Suppress AEL warnings on load. Reported by: eliel Patch by: eliel Closes issue #11178 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	37166d9a1a	Provide the ability to directly manipulate the TON/NPI bits in the dialstring. Reported by: thetatag Patch by: thetatag/stevens/tilghman Closes issue #5331 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	0cd3118a62	Adding the queue strategy wrandom (closes issue #10942, reported and patched by julianjm, documentation changes by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e9e78af981	Merged revisions 88994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88994 \| file \| 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) \| 6 lines Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b164d5a675	Add jitterbuffer support to chan_unistim. (closes issue #11168) Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	267683eb19	Merge the code from asterisk/team/group/chan_unistim: This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	e8c781b215	Add pbx_lua as a method of doing extensions Reported by: mnicholson Patch by: mnicholson Closes issue #11140 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	cf861b38c7	Added queue strategy "linear". This strategy is useful for those who always wish for their phones to be rung in a specific order. (closes issue #7279, reported and initially patched by diLLec, patch reworked by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	6cd5e1aee6	Remove information about the roundrobin strategy from trunk's queues.conf.sample since it no longer exists git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	a8cc80e36d	Adding the general option "shared_lastcall" to queues so that a member's wrapuptime may be used across multiple queues. (closes issue #9777, reported and patched by eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	0c14c47523	resetinterval defaulting to something other than 'never' doesn't seem to accomplish any good and causes problems for plenty of people... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Fredrickson	c5bb538818	Improved comments and organization for zapata.conf (#10904 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6998be1b3b	Document the changes made earlier today to meetme git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	cd1e6873aa	Merged revisions 86032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86032 \| mmichelson \| 2007-10-16 18:35:31 -0500 (Tue, 16 Oct 2007) \| 3 lines Since monitor-join is deprecated now, remove the example from the sample queues.conf file ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ed690fc348	Switch dundi to new tos config format. Remove old unused defines for old style. Closes issue 10860, patch by IgorG. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	fb9855eba1	Merged revisions 85571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85571 \| file \| 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) \| 4 lines Document that DTMF based features only work when two channels are bridged together. (closes issue #10773) Reported by: pbayley ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	fbcd884e1b	Allow for the position announcement to be turned off if desired. (closes issue #8515, reported by bruno_rocha, initial patch by bruno_rocha, final patch by qwell) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	510430a6a2	Make the status and priority configurable. Closes issue #10785, patch by Luke-Jr, thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	df30de142c	Add a new option for files-based music on hold to ensure that the sort order of the files is alphabetical. (closes issue #10855) Reported by: jamesgolovich Patches: asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license 176) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Dwayne M. Hubbard	0f53904918	merged jcmoore's patch for configurable SDP origin-field username and session field, closes issue# 10795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	0c8381a1f5	(closes issue #10739 ) Reported by: ruffle Patches: app_voicemail.c.diff uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by qwell (license 4) Tested by: callguy, ruffle Add an option to disable the automatic moving of "heard" messages to the Old folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	9a5f7c5764	(closes issue #10755 ) Reported by: snar Patches: app-queue-cdr-trunk.patch uploaded by snar (license 245) queues.conf.patch uploaded by snar (license 245) Add an updatecdr option to queues.conf, so that if a "member name" is specified, the cdr record will be updated with that, rather than the channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	a9c2f441d3	Merged revisions 82751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #10753) ........ r82751 \| qwell \| 2007-09-18 10:28:21 -0500 (Tue, 18 Sep 2007) \| 4 lines Correct the allowexternaldomains option in SIP sample config. Issue 10753 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	cb8c4122bc	Fix the sample redirect to point to a valid file in the Asterisk GUI. Closes issue #10748, patch by bkruse git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	da5930c234	Merged revisions 82435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82435 \| russell \| 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) \| 3 lines Add a note to help clarify the value set with the echocancel option. (inspired by Malcolm's blog post on blogs.digium.com about HPEC) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	4baba7c951	Add support in chan_skinny for sending RTP directly to the endpoints. Closes issue #9154, patch by DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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946 Commits (f82c42a22e6a0883c8b51b269d3be9c50e4cd6be)