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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines
Clarify limitonpeers=yes
(closes issue #11304)
Reported by: pj
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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Both still works in this version.
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added some verbage about the new algorithm to CHANGES.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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in light of commit 89441. Thanks to pj for pointing out the need for this
(closes issue #11307, reported by pj)
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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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
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(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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11195)
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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Reported by: thetatag
Patch by: thetatag/stevens/tilghman
Closes issue #5331
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(closes issue #10942, reported and patched by julianjm, documentation changes by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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(closes issue #11168)
Reported by: IgorG
Patches:
unistimjb-88863-1.patch uploaded by IgorG (license 20)
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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
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Reported by: mnicholson
Patch by: mnicholson
Closes issue #11140
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wish for their
phones to be rung in a specific order.
(closes issue #7279, reported and initially patched by diLLec, patch reworked by me)
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since it no longer exists
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wrapuptime
may be used across multiple queues.
(closes issue #9777, reported and patched by eliel)
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accomplish any good and causes problems for plenty of people...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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(closes issue #8515, reported by bruno_rocha, initial patch by bruno_rocha, final patch by qwell)
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Closes issue #10785, patch by Luke-Jr, thanks!
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of the files is alphabetical.
(closes issue #10855)
Reported by: jamesgolovich
Patches:
asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license 176)
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session field, closes issue# 10795
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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.
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #10753)
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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
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Closes issue #10748, patch by bkruse
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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Closes issue #9154, patch by DEA
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you can now use it with zaptel channels. (done while in Montreal at the Asterisk bootcamp!)
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(closes issue #10711, rbraun_proformatique)
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Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 Sep 2007) | 5 lines
Removing non-existent options from misdn configuration sample.
(closes issue #10678, reported and patched by IgorG)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep 2007) | 3 lines
Moving the explanation for joinempty to a more appropriate place
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