Age | Commit message (Collapse) | Author | Files | Lines |
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_cli_(un)register_multiple()
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files here.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39807 f38db490-d61c-443f-a65b-d21fe96a405b
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* fail on misdn_cfg_init() if elements in the config enum don't match with the config structs in misdn_config.c
* implemented first bits for encoding ISDN facility information elements via ASN.1 descriptions
* using unnamed semaphore for syncing in misdn_thread
* advanced fax detection: configurable detect timeout and context to jump into
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39378 f38db490-d61c-443f-a65b-d21fe96a405b
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* removed the holded element from the chan_list struct, we know this from the
state already
* added a few tweaks to make HOLD/RETRIEVE work again (TRANSFER does not work
yet)
* added possibility to debug mISDN frames via syslog
* added misdn_lib_port_is_blocked function to check if a port is blocked
* removed ec_training=1 from empty_bc, we don't use ec_training anymore
* removed unused misdn_lib_get_l2_status function
* added the nt bit to dummy misdn_bchannel objects
* setting bc->out_fac_type to FACILITY_NONE defaultly
* removed HANDLER_DEBUG stuff for better readability
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39295 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
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r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38905 f38db490-d61c-443f-a65b-d21fe96a405b
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* added blocking flag to stack object. A port can be blocked/unblocked from the
cli
* added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later
we can add a manager event for that)
* added block_on_alarm option, to block the port whenever a ALARM occurs
* added need_busy flag to indicate if we've sended a CONTROL_BUSY already
* changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in
recent asterisk ast_log messages..
* fixed a few ETSI state violations, especially when finishing calls in
different seldom states
* changed debug levels a lot to make the log more readable in low debuglevels
* some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still)
* removed the PRECONNECTED state stuff
* added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this
creates a CHANISUNAVAIL instead of a NOANSWER
* removed the addr pointer from "misdn show stacks" that's not needed anymore
and makes the output more unreadable
* added cause saving on RELEASE/RELEASE_COMPLETE
* set cause to 16 on prepare_bc
* removed stack getting from ph_control functions, we don't really need it
there
* added beroec api
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38801 f38db490-d61c-443f-a65b-d21fe96a405b
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37508 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@37382 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@37323 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36941 f38db490-d61c-443f-a65b-d21fe96a405b
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get a lot fewer blocked in thread blah warnings..
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36867 f38db490-d61c-443f-a65b-d21fe96a405b
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settings with a description in the CLI
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36865 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36379 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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debug messages to higher log level
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* fixed tone handling after ast_hangup was called
* optimized the tone_indication function
* removed warnings in favour of log debugs
* improved the round_robin method
* added logs for channel setting/emptying
* fixed channel forgot to set bug
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36082 f38db490-d61c-443f-a65b-d21fe96a405b
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* fixed a few inband Alerting issues, sometimes we need to create alerting, some
times it's inband
* beautified the state debugging of misdn_hangup
* removed "real" bchannel activating/deactivating in chan_misdn.c
* fixed "round_robin" bug when there's only 1 port
* added more informative prints when channel could not be created
* changed some warnings to notices
* reworked the whole bchannel state machine stuff,
it is now like in the examples of mISDNuser and therefore a lot easier,
and it is now harder to create bugs
* bchannel_activate/deactivate is now only called in setup/cleanup bc,
they may merge sometime
* it is very important to setup/cleanup the bchannels under the correct
conditions, especially in the NT Side we can only setup the bchannels
when we send a Message!
In the TE side we can only setup the bchannel when we received the channel
of course
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tones, therefore the TONE_X defines are in the isdn_lib.h now. changed a REALEASE to a DISCONNECT in NT Stack, to make tones available in that state.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34604 f38db490-d61c-443f-a65b-d21fe96a405b
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* added early bridge-hook, so we know if we need to generate ringing or
can take it from the far end chan_misdn channel (if available)
* fixed the issue, that we may not activate the bchannel on PTMP,
when we receive ALERTING/PROCEEDING/PROGRESS, only on CONNECT. There might
be other PTMP devices and we might disturb their bchannel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34552 f38db490-d61c-443f-a65b-d21fe96a405b
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again :-)
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the conference nr. when bridged
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defines for deadlock debugging. added code snippet for generating silence if we don't have data to write.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32524 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31324 f38db490-d61c-443f-a65b-d21fe96a405b
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channel, also changed name generation to new stringfield api
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30132 f38db490-d61c-443f-a65b-d21fe96a405b
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transmit of progress indicators through channel vars
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29938 f38db490-d61c-443f-a65b-d21fe96a405b
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the l1state.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29803 f38db490-d61c-443f-a65b-d21fe96a405b
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* when receiving a connect from the NT Side we wait until we have the final
l3id until we queue the answer to asterisk to avoid bridging conflicts
* when not bridged to misdn we had a segfault after receiving the connect
due to a strcasecmp bug.. this didn't happen before, cause we hadn't had
the bridge before
* cleanup of the bchannels is queued now, due to possible race conditions
* added mISDN_clear_stack when cleaning the bchannel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29667 f38db490-d61c-443f-a65b-d21fe96a405b
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disconnect indications again
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27770 f38db490-d61c-443f-a65b-d21fe96a405b
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removed redundant buffer betweend mISDN reading thread and ast_read in favour of the already existing pipe, this clarifies the way a voice frame takes between mISDN and asterisk a lot. centralized debugging of NumberPlan. removed a compiler warning.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27346 f38db490-d61c-443f-a65b-d21fe96a405b
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Ringing on other channels if they feel that they should have inband ringing, but there is non in reality. I need this due to the fact that asterisk has not the possibility to transmit progress indicators thus chan_sip and others do not know wether they should generate a Rining tone themselves if they receive AST_CONTROL_RINGING..
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attempt to free() them later)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23854 f38db490-d61c-443f-a65b-d21fe96a405b
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also changed some strdups to ast_strdupa
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23443 f38db490-d61c-443f-a65b-d21fe96a405b
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