Age | Commit message (Collapse) | Author |
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and fix a couple little things in passing
- usecnt was not initialized in chan_iax2
- ast_update_use_count() was not called after incrementing the count in chan_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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if it's the "right" fix. Please review and make any adjustments you see necessary.
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inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
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exists for it... otherwise the default values will cause codec combination madness
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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37949 | kpfleming | 2006-07-19 12:10:10 -0500 (Wed, 19 Jul 2006) | 2 lines
ensure that global 'maxauthreq' is reset to zero during 'reload'
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don't reuse LIBS variable from top-level Makefile (oops)
build Asterisk binary after subdirs (preparing for embedded modules)
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needed (although none do today)
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connected,
in vm_authenticate, it was try to start ADSI on the channel, and it WOULD because it
was "supported", according to the iaxy. There is now a config option (adsi=yes) for
this, which defaults to no. (config sample coming shortly)
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Abstract out some common code into a single function
With the recent scheduler "issues", it pointed out a few things I might have been missing,
so I added some rudimentary vrtp and rtcp stuff
General cleanup...
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37531 | kpfleming | 2006-07-13 11:44:23 -0500 (Thu, 13 Jul 2006) | 2 lines
report address of peer trying to subscribe to unknown hint
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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been requested (Sipura/Grandstream ATAs and others will need this)
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split support for G726-32 into RFC3551 and AAL2 packing orders, since both are in use
change "G726-32" to be RFC3551 packing order, in spite of devices that use AAL2 order with this MIME type
add ability to directly transcode between packing orders
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I WILL TYPE "make" BEFORE COMMITTING ANY CODE
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Bug found in SIP Master Class, Chicago)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37439 | file | 2006-07-12 11:23:59 -0400 (Wed, 12 Jul 2006) | 2 lines
Add support to have maxauthreq as a global option
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37419 | kpfleming | 2006-07-12 08:54:10 -0500 (Wed, 12 Jul 2006) | 2 lines
remove some more bad examples of using printf
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37417 | kpfleming | 2006-07-12 08:18:21 -0500 (Wed, 12 Jul 2006) | 2 lines
get rid of some more printf's (although most of these were ifdef-ed out)
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Now l1watcher timeouts can be configured separately for every portgroup.
* added a signal handler to allow waking up the misdn task thread (that may sleep in a poll call) via misdn_tasks_wakeup().
* overlap_dial functionality implemented.
* fixes a bug which leads to a segfault after reordering config elements in the enum or struct
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
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find_chan_by_l3id, since the l3id is not unique over all ports. removed automatic nt_stack reinitialization, since this creates segfaults.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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commands, forward hold/unhold in dial, add hold device state
and implement holding in the SLA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37212 | bweschke | 2006-07-06 15:38:45 -0500 (Thu, 06 Jul 2006) | 3 lines
Don't do wierd things on a callback agent that has attempted logoff while still on the phone.
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NOISY_BUILD variable in the top-level Makefile)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37173 | file | 2006-07-06 11:48:07 -0400 (Thu, 06 Jul 2006) | 2 lines
Instead of giving the scheduled item ID on a peer expiration, give the time until they expire (issue #7455 reported by slavon)
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r36998 | file | 2006-07-05 11:31:01 -0400 (Wed, 05 Jul 2006) | 2 lines
Spell extension correctly in documentation for chan_oss dial (issue #7487 reported by flefoll)
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only set it when loading chan_misdn for now. fixed a litle state problem when receiving RELEASE_COMPLETE. also we may only play tones to a NT when the extension does not match and such cases.
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get a lot fewer blocked in thread blah warnings..
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settings with a description in the CLI
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route headers, we're reaching the limimt for what's possible in one UDP packet...
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