Russell Bryant
87ac16847e
- unregister SLA apps on module unload and add sample config (issue #7701 , junky)
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- rename SLAS to SLAStation, and SLAT to SLATrunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-14 01:32:46 +00:00
Joshua Colp
a0bd41f79b
Add support for Sigma Designs cards. These basically allow you to offload dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 17:02:45 +00:00
Joshua Colp
0f0323dbab
Clarify volgain option a bit, it needs sox to work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 15:53:51 +00:00
Tilghman Lesher
f37a4e3e12
Bug 6237 - add volgain parameter, such that voicemail messages may be amplified after recording
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 15:37:53 +00:00
Kevin P. Fleming
9d26f46fc7
remove some extraneous 'followme' in prompt names
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07 18:47:33 +00:00
Olle Johansson
2f69bec40e
Add placeholder for sla.conf sample in configs/. Please update with
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07 17:01:40 +00:00
Russell Bryant
4d7c67fc72
Merge my applicationmap_fixup branch to address the issues described in this
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post to the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html
This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on. I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-07 04:15:52 +00:00
Tilghman Lesher
0d902b3033
Update documentation on realtime; add a workaround for lack of realtime hints by using func_odbc
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-03 19:05:06 +00:00
Kevin P. Fleming
6d0742fc16
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19 20:44:39 +00:00
Mark Spencer
837910062b
First pass at in-place file manipulation via manager
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19 13:28:38 +00:00
North Antara
e69e056012
config sample for the previous, regarding ADSI
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-14 21:49:40 +00:00
Kevin P. Fleming
4376af0080
actually make the non-standard G726-32 behavior available for SIP clients
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13 20:35:41 +00:00
Christian Richter
54ce0f0a22
added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13 14:13:24 +00:00
Olle Johansson
3b5a2aafa4
- Change filename to current file name
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-12 20:56:49 +00:00
Joshua Colp
ba092c1244
And now the trunk version! Add an option for IAX2 users that allows you to set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-11 19:03:56 +00:00
Olle Johansson
0e0059c0f3
Remove configuration option "restrictcid" that is nowhere to
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be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-10 11:20:49 +00:00
Matthew Fredrickson
de03118578
Asterisk portion of the T309 patch. ( #7271 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-06 21:51:51 +00:00
Christian Richter
bd0b801a0d
* removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
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* removed the state handling from release_chan, and simplified the ast_hangup/ast_queue_hangup stuff
* added pp_l2_check option, for pp lines where the pbx does not initially gets the L2 up
* simplified and fixed a bug in the pid generation code
* fixed a bug in empty_chan, which might cause segfaults and memorry corruptions
* added prepare_bc function, which is sort of the opposite of empty_bc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-06 15:11:40 +00:00
Olle Johansson
ec9d4711d7
- Add notes about voicemail depending on res_adsi
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- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 06:06:23 +00:00
Olle Johansson
b971f65978
- Make use of system name in realtime SIP peers optional
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- Fix small issue with SIP history
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-02 12:00:36 +00:00
Olle Johansson
f3594bd1a0
Removing configuration options that does not do anything yet. No need to
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add "promises" to the sip.conf.sample...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-30 07:18:30 +00:00
Christian Richter
f5c0cd2ddc
added better L2 handling for ptp, if it's down we don't try to call on that port in groupdial anymore, also we try to get it up then. Additionally added the configoptions ntdebugflags and ntdebugfile to debug the mISDNuser NT Stack (should have done that ages before..). isdn_lib.c compiles again.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 20:12:19 +00:00
Kevin P. Fleming
dec3d7d4c0
Merged revisions 36253-36254 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines
add documentation for peer-specific 'outboundproxy' setting
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r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines
clarify documentation for 'persistentmembers' setting
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 08:01:08 +00:00
Olle Johansson
4177596e8d
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 07:04:43 +00:00
Olle Johansson
cc43f0bdc7
Speling error. Avoid swenglish :-) (thanks, jtodd!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 18:34:29 +00:00
Olle Johansson
6399ac438d
Add explanation and warning about the "s" extension. (Hi Mike :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 17:31:51 +00:00
Olle Johansson
f8311adcda
METERMAIDS:
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-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 16:43:21 +00:00
Olle Johansson
e2b0c5b558
Add example of permit/deny to sip.conf.sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 16:24:43 +00:00
Tilghman Lesher
89f6ffe1e5
Bug 6589 - option to display channel variables in queue events
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-22 15:43:02 +00:00
Joshua Colp
4c066de7cf
Merged revisions 35334 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r35334 | file | 2006-06-21 16:25:34 -0300 (Wed, 21 Jun 2006) | 2 lines
Add Venezuelan indications (issue #7402 reported by palillo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-21 19:26:52 +00:00
North Antara
a5d6979fac
Finally merge chan_skinny fixes into trunk.
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ready...set...GO!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-18 22:36:39 +00:00
Russell Bryant
035a8b4278
Merged revisions 34627 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r34627 | russell | 2006-06-18 16:15:15 -0400 (Sun, 18 Jun 2006) | 5 lines
don't store multiple secrets delimited with semicolons for peers because this
is only valid for users. Instead, only keep the last specified secret for a
peer entry. Also, document how multiple secrets are handled in the sample
config. (Reported by PCadach on #asterisk-bugs)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-18 20:18:41 +00:00
Joshua Colp
0e5e744fb2
Add bulgarian indications (issue #7314 reported by KNK)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-16 16:23:13 +00:00
Joshua Colp
5456f425c6
Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-13 19:38:41 +00:00
Matt O'Gorman
1e530787f3
solves some bugs with memory allocation, and adds
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a message stack.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-07 22:43:20 +00:00
Kevin P. Fleming
bc49d5bfb3
moh files will now be distributed in native format, not mp3, so...
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update sounds/Makefile to download/unpack the proper files
remove all evidence of mpg123 from the main Makefile
change the sample musiconhold.conf file to use native mode instead of mp3
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-07 22:37:31 +00:00
BJ Weschke
3d973a0686
Introducing app_followme into /trunk!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-06 07:29:57 +00:00
Kevin P. Fleming
dd6de5ee4e
it's time... only enable global priority jumping if the config file says to do so
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-05 14:00:13 +00:00
Olle Johansson
4e74b8fb75
Issue #7231 - Missing indications from libtonezone (tzafrir)
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- "tuttter" changed to "stutter" ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-05 09:27:38 +00:00
Olle Johansson
2dc6947144
Issue #2863 - Improved RTCP support (John Martin, Fredrik Olsson)
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Thanks to everyone involved in working with this!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-05 08:39:42 +00:00
Matt O'Gorman
18b135f215
oops my config file was out of date not the sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-04 17:08:18 +00:00
Matt O'Gorman
58cd5f2440
oops fixing example config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-04 17:07:09 +00:00
Russell Bryant
4c76028de9
- add the ability to configure forced jitterbuffers on h323, jingle,
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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257 , north, with additional updates and modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 16:47:28 +00:00
Christian Richter
4be235a974
added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 12:51:41 +00:00
Kevin P. Fleming
6bce269454
Merged revisions 31321 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines
remove a sample entry that never should have been added (code to support it was not merged)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 12:43:01 +00:00
Matt O'Gorman
8cfb992c1e
adds statusmessage customization from Julian Lyndon-Smith
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and fixes bug with pruneregister
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 07:49:47 +00:00
North Antara
e25c4621b4
Nobody saw this coming, I bet.
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jitterbuffer!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 04:35:00 +00:00
Russell Bryant
bb7dd96cfe
Add support for using a jitterbuffer for RTP on bridged calls. This includes
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854 , Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 16:56:50 +00:00
Kevin P. Fleming
18606233da
fix various typos and other bits (from Ian Kinner)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-30 16:01:50 +00:00
Joshua Colp
18248f092f
Merged revisions 30239 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r30239 | file | 2006-05-25 12:27:44 -0300 (Thu, 25 May 2006) | 2 lines
Get rid of an incorrect SIP dial string in the sample extensions.conf - I even tried variations... no go (issue #7222 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-25 15:29:30 +00:00