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
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
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
- 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
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
* 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: http://svn.digium.com/svn/asterisk/trunk@89470 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89441 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89250 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89073 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89069 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@88368 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@85682 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82257 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: jmls
Patches:
pbx.diff uploaded by jmls (license 141)
Add REASON dialplan variable for when an originated call fails and the failed extension is executed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81372 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81233 f38db490-d61c-443f-a65b-d21fe96a405b
oej