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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: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@97203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@95412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@95167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@94799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@94782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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queue-penalty branch
was the CHANGES file. No longer!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@93854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@93166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@93165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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date,
even with experimental stuff.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: eliel
Patch by: eliel
(Closes issue #11344)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@91347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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"call" level.
(Closes issue #11015)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Closes issue #11464, patch by eliel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: eliel
Patches:
core.show.hint.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: eliel
Patches:
CHANGES.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@90656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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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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Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
(patched by me, and tested internally)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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with the SIPPEER() argument of the same name. The deprecation procedure is not
being used here since this is a trunk-only option.
(closes issue #11307, reported by pj, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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parked extension is provided as an argument.
(closes issue #10803)
Reported by: outtolunc
Patches:
res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237)
- modified by me to work a bit differently ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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AGI debugging on busy systems much easier.
(closes issue #10730)
Reported by: junky
Patches:
agi_debug_chan.diff uploaded by junky (license 177)
20070923_10730.diff uploaded by mvanbaak (license 7)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(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
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much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Thank you Russell, for pointing this out!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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It allows you to configure a prefix for auto-monitor recordings.
(closes issue #6353)
Reported by: ivanfm
Patches:
asterisk_automon_v4.patch uploaded by ivanfm (original patch)
- updated patch:
6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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you can now use it with zaptel channels. (done while in Montreal at the Asterisk bootcamp!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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