Commit Graph

204 Commits

Author SHA1 Message Date
Olle Johansson 25cbb792b9 (closes issue #11422)
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
2007-12-04 15:07:53 +00:00
Olle Johansson d5c7e96526 (closes issue #11462)
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
2007-12-04 15:02:48 +00:00
Joshua Colp 8bfdea3160 Add AGI commands for speech recognition. These mirror the dialplan 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
2007-12-03 21:03:05 +00:00
Mark Michelson a42259c3ff Adding support for realtime music on hold. The following are the main points:
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
2007-11-28 00:47:22 +00:00
Olle Johansson 130a2051fa - Mark "concise" as deprecated
- 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
2007-11-26 19:24:23 +00:00
Steve Murphy 2ec4b57622 Thanks to pnlarsson for noting the spelling error in the cli commands. Also, 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
2007-11-26 16:24:27 +00:00
Olle Johansson 07cb09ad86 - Deprecate "call-limit" in chan_sip. No other channel driver enforces 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
2007-11-25 11:46:17 +00:00
Tilghman Lesher 1c295be7a0 Change Read to set READSTATUS as an indication of the result
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
2007-11-21 18:38:18 +00:00
Russell Bryant 6335b4b30d Merge changes from team/russell/sla_trunk_moh ...
* 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
2007-11-21 00:21:38 +00:00
Mark Michelson fb3b4f4937 Changed the "busy-level" option in sip.conf to "busylevel" to be more parallel
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
2007-11-19 23:24:35 +00:00
Mark Michelson 67f044d42a Adding SYSINFO() dialplan function for retrieval of system information
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 16:29:07 +00:00
Olle Johansson 19014f31d9 Update CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 09:16:56 +00:00
Russell Bryant fa39f74761 Update the ParkedCall application to grab the first available parked call if no
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
2007-11-13 20:30:13 +00:00
Russell Bryant 4afb905cf0 Print out the channel name as a prefix to the "agi debug" output. This makes
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
2007-11-07 00:00:38 +00:00
Russell Bryant e309393920 Added the ability to do "meetme concise" with the "meetme" CLI command.
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
2007-11-06 23:44:39 +00:00
Mark Michelson 0cd3118a62 Adding the queue strategy wrandom
(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
2007-11-06 22:36:55 +00:00
Russell Bryant a06218ee6d Added the S() and L() options to the MeetMe application. These are pretty
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
2007-11-06 22:15:32 +00:00
Tilghman Lesher 00ad9612be Change wording to that suggested by MasterYoda
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05 18:22:20 +00:00
Russell Bryant 267683eb19 Merge the code from asterisk/team/group/chan_unistim:
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
2007-11-02 20:56:12 +00:00
Tilghman Lesher a6fb1baef0 Add a few bytes on LUA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02 16:26:31 +00:00
Mark Michelson a55b6954e8 Forgot to update CHANGES when I committed the linear queue strategy.
Thank you Russell, for pointing this out!



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-26 22:21:08 +00:00
Tilghman Lesher 6998be1b3b Document the changes made earlier today to meetme
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-17 20:42:20 +00:00
Russell Bryant ea02f3d0c5 Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
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
2007-10-15 20:08:04 +00:00
Russell Bryant 5aaaaed28d Note jitterbuffer support for chan_local in CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09 15:12:59 +00:00
Mark Michelson eb39b71fba Added the ability to pause and unpause members via the CLI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13 21:23:32 +00:00
Joshua Colp 5460e72015 Add setvar support to chan_zap. Just like you can in chan_sip and chan_iax2 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
2007-09-13 16:58:59 +00:00
Joshua Colp 9642d93117 (closes issue #9433)
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: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-11 17:58:48 +00:00
Russell Bryant b068a17e60 Add EXTENSION_STATE() function that can retrieve the state of an extension that
has a hint.

(closes issue #10635, adamgundy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 20:54:07 +00:00
Russell Bryant 905f15d0b0 s/DEVSTATE/DEVICE_STATE/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 20:27:53 +00:00
Russell Bryant 65b4a88c60 Merge HINT() dialplan function from my sandbox branch into trunk. This function
will let you retrieve the list of devices or name associated with a hint.
(inspired by issue #10635)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 20:24:18 +00:00
Joshua Colp f614bc7004 (closes issue #10377)
Reported by: mvanbaak
Patches:
      chan_skinny_info.diff uploaded by mvanbaak (license 7)
Add skinny show device, skinny show line, and skinny show settings CLI commands.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 20:16:02 +00:00
Joshua Colp 56e74f0dde (closes issue #10603)
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: https://origsvn.digium.com/svn/asterisk/trunk@81372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-30 14:42:41 +00:00
Russell Bryant 43e9b0f67c (closes issue #7852)
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: https://origsvn.digium.com/svn/asterisk/trunk@81233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-28 16:28:26 +00:00
Olle Johansson 0c321a54d9 Doc change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16 06:52:17 +00:00
Steve Murphy 9836efb5fb This commit closes bug 7605, and half-closes 7638. The AEL code has been redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-15 19:21:27 +00:00
Mark Michelson 8d929d7afd Allow non-realtime queues to have realtime members
(issue #10424, reported and patched by irroot)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13 15:39:48 +00:00
Tilghman Lesher 3257acb922 Add some documentation detailing an aspect of dialplan functions, as requested by Russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-31 18:50:06 +00:00
Russell Bryant de1bcbc423 remove a couple of entries that got duplicated and snuck into the SIP section. Also, align the NAT/STUN entry with the others.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-25 01:06:02 +00:00
Luigi Rizzo 5305d61e85 add documentation on nat/stun support in chan_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-24 07:51:14 +00:00
Russell Bryant 098acf6fc3 note the debug and verbose changes in CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-23 14:23:47 +00:00
Olle Johansson 22bb315824 Update with new features
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-09 08:30:04 +00:00
Russell Bryant 8c598f0e11 Redistribute a lot of the items that were in the Misc. section
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-06 03:48:33 +00:00
Russell Bryant 98b08197f3 note TLS support for manager and HTTP in CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-06 03:40:57 +00:00
Joshua Colp 62084eb2a4 Add SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables when a transfer takes place. (issue #8378 reported by jcovert)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 23:13:09 +00:00
Mark Michelson 5310385315 Added ability to customize which buttons control forward, reverse, pause, and stop during message playback.
(closes issue 9474, reported and patched by jaroth with modifications by me)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 22:47:08 +00:00
Mark Michelson 4596af13fc Adding feature to support the storage and retrieval of voicemail greetings using IMAP storage.
This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf
voicemail.conf.sample has details on the options added.

As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined.

In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders
and so now the code should be easier to understand and maintain when it comes to this area.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 19:50:21 +00:00
Joshua Colp 1961b57705 Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-26 23:31:23 +00:00
Steve Murphy c1bb0fc34b This finishes the changes for making Macro args LOCAL to the call, and allowing users to declare local variables.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20 20:10:19 +00:00
Steve Murphy 75e6a8f807 Added a little verbage to CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19 23:38:54 +00:00
Steve Murphy abf614c5a1 Moved those comments from UPGRADE.txt to CHANGES. Ooops.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19 21:58:51 +00:00