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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40392 | kpfleming | 2006-08-18 14:10:51 -0500 (Fri, 18 Aug 2006) | 2 lines
make a feeble attempt to avoid the 'how do I enable my hardware echo canceler' questions
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junky)
- rename SLAS to SLAStation, and SLAT to SLATrunk
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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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amplified after recording
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39182 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
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hints by using func_odbc
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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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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set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37380 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
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hanguptone of course)
* 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: http://svn.digium.com/svn/asterisk/trunk@37172 f38db490-d61c-443f-a65b-d21fe96a405b
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- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36979 f38db490-d61c-443f-a65b-d21fe96a405b
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- Fix small issue with SIP history
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add "promises" to the sip.conf.sample...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36355 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36298 f38db490-d61c-443f-a65b-d21fe96a405b
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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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-----------
- 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
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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: http://svn.digium.com/svn/asterisk/trunk@35335 f38db490-d61c-443f-a65b-d21fe96a405b
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ready...set...GO!
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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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a message stack.
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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
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do so
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- "tuttter" changed to "stutter" ;-)
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Thanks to everyone involved in working with this!
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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)
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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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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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and fixes bug with pruneregister
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31275 f38db490-d61c-443f-a65b-d21fe96a405b
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jitterbuffer!
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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!
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