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https://origsvn.digium.com/svn/asterisk/trunk
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r130044 | kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 lines
clean up a bunch more Zaptel-related references
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https://origsvn.digium.com/svn/asterisk/trunk
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r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 lines
Merged revisions 122127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
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https://origsvn.digium.com/svn/asterisk/trunk
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r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, 12 Jun 2008) | 45 lines
Merged revisions 122046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
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https://origsvn.digium.com/svn/asterisk/trunk
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r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines
Add native AGI command GOSUB, as invoking Gosub with EXEC does not work
properly.
(closes issue #12760)
Reported by: Corydon76
Patches:
20080530__bug12760.diff.txt uploaded by Corydon76 (license 14)
Tested by: tim_ringenbach, Corydon76
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https://origsvn.digium.com/svn/asterisk/trunk
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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines
Merged revisions 118646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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https://origsvn.digium.com/svn/asterisk/trunk
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r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines
Merged revisions 110880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines
due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves
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https://origsvn.digium.com/svn/asterisk/trunk
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r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines
Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues.
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do some additional code cleanup and improvement in passing.
(closes issue #12106)
Reported by: nizon
Patches:
devstate-patch.txt uploaded by nizon (license 415)
-- Updated to trunk, and tab completion added by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105461 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11170)
Reported by: kratzers
Patches:
ResetCDR.1.diff uploaded by kratzers (license 307)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104215 f38db490-d61c-443f-a65b-d21fe96a405b
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changes are backward-compatible, so no changes to UPGRADE.txt are
necessary.
(closes issue #9279)
Reported by: rottenroddy
Patches:
20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104101 f38db490-d61c-443f-a65b-d21fe96a405b
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privilege to call out to a subshell.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104039 f38db490-d61c-443f-a65b-d21fe96a405b
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application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked.
(closes issue #11553)
Reported by: johan
Patches:
UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
CHANGES.channelredirect.patch uploaded by johan (license 334)
app_channelredirect-20080219.patch uploaded by johan (license 334)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103819 f38db490-d61c-443f-a65b-d21fe96a405b
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- Add new event to CHANGES
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103755 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11268)
Reported by: moy
Patches:
chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103754 f38db490-d61c-443f-a65b-d21fe96a405b
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Added ability to retrieve list of categories in a config file.
Added ability to retrieve the content of a particular category.
Added ability to empty a context.
Created new action to create a new file.
Updated delete action to allow deletion by line number with respect to category.
Added new action insert to add new variable to category at specified line.
Updated action newcat to allow new category to be inserted in file above another existing category.
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channel if it has not already been answered.
(closes SPD-51)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101082 f38db490-d61c-443f-a65b-d21fe96a405b
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drops packets that do not come from the remote party.
(closes issue #8952)
Reported by: amorsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100206 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100039 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #5768)
Reported by: mguesdon
Patches:
res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121)
res_ldap.conf.sample uploaded by suretec (license 70)
asterisk-v3.1.4.ldif uploaded by suretec (license 70)
asterisk-v3.1.4.schema uploaded by suretec (license 70)
Tested by: oej, mguesdon, suretec, cthorner
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99696 f38db490-d61c-443f-a65b-d21fe96a405b
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from cli.conf
after a discussion on the -dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99642 f38db490-d61c-443f-a65b-d21fe96a405b
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
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immediately at startup. Any commands in the startup_commands file in the Asterisk
config diretory will get executed.
(closes issue #11781)
Reported by: jamesgolovich
Patches:
asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
-- With some changes by me.
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prodding by jsmith.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98969 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10740)
Reported by: ruffle
Patches:
app_voicemail.diff uploaded by ruffle (license 201)
10740-voicemail.diff uploaded by qwell (license 4)
20080113_bug10740.diff.txt uploaded by mvanbaak (license 7)
Tested by: blitzrage, mvanbaak, qwell
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- Change spacing a bit in some places for consistent indentation
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98656 f38db490-d61c-443f-a65b-d21fe96a405b
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Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
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variable (or function) on an active channel from the CLI.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98558 f38db490-d61c-443f-a65b-d21fe96a405b
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gets generated.
(closes issue #11212)
Reported by: tzafrir
Patches:
zap_dnd.diff uploaded by tzafrir (modified by me) (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98488 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11690)
Reported by: tzafrir
Patches:
signaling_to_signalling.diff uploaded by tzafrir (license 46)
signalling_cleanup.diff uploaded by tzafrir (license 46)
zap_auto_default.diff uploaded by tzafrir (license 46)
zap_no_default_sig.diff uploaded by tzafrir (license 46)
zap_signal_auto.diff uploaded by tzafrir (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98436 f38db490-d61c-443f-a65b-d21fe96a405b
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to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98027 f38db490-d61c-443f-a65b-d21fe96a405b
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
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based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
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for a queue. app_queue
will monitor this second device's state for the member, even though it actually calls the first
interface. This ability has been added for statically defined queue members, realtime queue members,
and dynamic queue members added through the CLI, dialplan, or manager.
(closes issue #11603, reported by acidv)
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This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
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Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
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The channel name is printed in verbose messages
maximumWordLength option added.
Duration of words that do not meet the minimum word duration will be logged
The duration of pre-greeting silence will be logged
Only consider us in the greeting if we actually detected a valid word duration.
(closes issue #11650, reported and patched by davevg)
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the existence of a dialplan target.
(closes issue #11579)
Reported by: irroot
Patches:
func_dialplan2.c uploaded by irroot (license 52)
-- Additional changes by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94799 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11625, reported and patched by sergee)
Thank you very much to sergee for adding this new feature!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94782 f38db490-d61c-443f-a65b-d21fe96a405b
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queue-penalty branch
was the CHANGES file. No longer!
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