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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99647 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98986 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98984 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98889 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98811 f38db490-d61c-443f-a65b-d21fe96a405b
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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 ...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98628 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97203 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95839 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95167 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94902 f38db490-d61c-443f-a65b-d21fe96a405b
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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!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94546 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93899 f38db490-d61c-443f-a65b-d21fe96a405b
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by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
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instead of pointing
to configuration file with -C
Reported by: sobomax
Patches:
asterisk.c.diff.trunk uploaded by sobomax (license 359)
doc changes by committer
(closes issue #11598)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93854 f38db490-d61c-443f-a65b-d21fe96a405b
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you need to reload after changes. Thanks YS.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(related to issue #11414)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93166 f38db490-d61c-443f-a65b-d21fe96a405b
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for a
manager realtime implementation.
If you change accounts in manager.conf, you now need to reload to activate the
changes (deletions, additions). This was not the case with 1.4.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(closes issue #11414)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93165 f38db490-d61c-443f-a65b-d21fe96a405b
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date,
even with experimental stuff.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93164 f38db490-d61c-443f-a65b-d21fe96a405b
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93160 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: eliel
Patch by: eliel
(Closes issue #11344)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93066 f38db490-d61c-443f-a65b-d21fe96a405b
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Thanks, eliel, for writing the original patch. Modified by me to follow
other manager events and the new "moremanager" style.
(closes issue #11478)
Reported by: eliel
Patches:
manager.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91347 f38db490-d61c-443f-a65b-d21fe96a405b
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"call" level.
(Closes issue #11015)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91172 f38db490-d61c-443f-a65b-d21fe96a405b
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Closes issue #11464, patch by eliel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90991 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@90949 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: eliel
Patches:
core.show.hint.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90853 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: eliel
Patches:
CHANGES.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90852 f38db490-d61c-443f-a65b-d21fe96a405b
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applications mostly but present the information in a nicer fashion. The SPEECH RECOGNIZE command for example will return the results instead of having to query the dialplan functions.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90656 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89946 f38db490-d61c-443f-a65b-d21fe96a405b
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- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
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added some verbage about the new algorithm to CHANGES.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89583 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89489 f38db490-d61c-443f-a65b-d21fe96a405b
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