channel to remove the theoretical race condition that the channel could get
bridged before the channel's jitterbuffer gets configured. This was pointed
out by PCadach on IRC. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@39378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@39295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r39081 | russell | 2006-08-06 21:28:29 -0400 (Sun, 06 Aug 2006) | 7 lines
Fix a crash reported to me by hads on IRC. This crash would occur with the use
of the "distinctiveringaftercid" option. Also, on this user's system, the crash
would only occur when built without optimizations. This is because the bug is
that the code would write past the end of an array that was allocated on the
stack, and the structure of the stack is different with or without optimizations
enabled.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- use appropriate types in some assignments
- use ast_strlen_zero()
- don't manually free cid fields since ast_set_callerid() will handle it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
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
........
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()
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@38801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38611 | kpfleming | 2006-07-31 16:14:11 -0500 (Mon, 31 Jul 2006) | 4 lines
don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough)
don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others)
use ast_softhangup_nolock() since we already hold the channel's lock
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
int to string or string to int operations.
"pure" essentially says that this function has no side effects aside from its
result, and the result depends on nothing else other than its arguments and
global variables. "const" is a more strict form of "pure", where the function
also doesn't access any global variables.
From the gcc manual: "Such a function can be subject to common subexpression
elimination and loop optimization just as an arithmetic operator would be."
This also tells the compiler that it is safe to call the function fewer times
than the code says to, given the same arguments, since the result will always
be the same.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38420 | file | 2006-07-28 14:49:00 -0400 (Fri, 28 Jul 2006) | 2 lines
Make a copy of the request URI in check_user_full instead of modifying the one on the structure, and also strip params properly from the user portion of the SIP URI so as to preserve the domain (issue #7552 reported by dan42)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38328 | russell | 2006-07-27 00:25:41 -0400 (Thu, 27 Jul 2006) | 4 lines
Fix crash when using the "regexten" option with MALLOC_DEBUG enabled. This was
not reported in the bug tracker but the same bug has been demonstrated in other
places in the code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
recent hold changes so that MOH is not started on the bridged channel directly.
However, the change is still not a bad idea.
Merged revisions 38200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38200 | russell | 2006-07-25 15:43:38 -0400 (Tue, 25 Jul 2006) | 6 lines
This resolves a deadlock that a tech support customer was getting frequently
when his users would answer call waiting. If another thread is currently
holding the zt_pvt lock for the first channel, unlock both channels and let
asterisk retry the native bridge, just like what is done for the second channel
directly below these changes.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
* 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
........
r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines
use ast_set_callerid to be more consistent and to make sure that the
"callerid" option in the conf files is always handled the same way and sets ANI
(issue #7285, gkloepfer)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- we need to investigate support for refusing offers of 30 ms (like the Nokia
e-series) or supporting it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in RFCs, as they keep referring to each other in a circular pattern in regards
to this item, but since the Nokia SIP/GSM phones use this, we might as well
start supporting it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@36082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
support the new location for zaptel.h and tonezone.h
use the dependency information output by menuselect to build Makefile rules for each module for header files and libraries
combine the common rules into a top-level Makefile.rules file
remove all (now) unnecessary stuff from subdir Makefiles
change translator API so that the newpvt() callback returns an int instead of a pointer (it no longer allocates memory)
alphabetize --with-<foo> options in configure script
enhance Net-SNMP support in configure script to provide a --with-netsnmp option
fix support for --with-pq so that if pg-config is not found when --with-pq is specified, an error will be generated
add 'optional package' usage to modules now that menuselect can output it
allow res_snmp to build by default, since the new loader changes coming soon will solve the function naming problem (and users can disable it via menuselect anyway)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- don't call the byte swapping macros on single byte numbers
- don't do a ++ increment in the argument in the argument to the byte swapping
macros. This gets expanded to incrementing the variable 4 times in a single
operation, which results in undefined (and obviously undesired) behavior. :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allocated. These changes caused crashes when using a channel type that did
not support the jitterbuffer. Instead of fixing why it's crashing, I'm going
to implement this in a better way next week. The way I did it caused a
jitterbuffer to be allocated on every channel where the channel type supported
jitterbuffers, even if they were disabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- if the pbx fails to start, set the owner channel of the pvt strucutre
to be NULL
- return immediately if the pbx fails to start so the loop to set all of
the variables from the "setvar" options aren't set as a bunch of global
variables instead
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
from configuring a jitterbuffer on a new channel because of a memory
allocation error
- On passing through these channel drivers, configure the jitterbuffer before
starting the PBX thread instead of afterwards. If the pbx fails to start for
whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
NULL in failure conditions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
subdirectory instead of a for loop
- remove the FORCE target from the main Makefile and add the couple places
I used it to the .PHONY target. .PHONY does the same thing and is a built-in
more efficient way of doing it.
- add a bunch more targets to .PHONY ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- add a copyright header to the build_tools Makefile
- remove 'depend' from the 'all' target in agi/ and utils/ since it is handled
by the main Makefile already
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
since they are targets that do not have resulting files and are never listed
as prerequisites to real targets. Using .PHONY in this manner improves make
performance by never having to check for resulting files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
More ideas for developing better video support in Asterisk?
Join the asterisk-video mailing list to help out in the
Asterisk Video Task Force!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Nadi Sarrar
Russell Bryant
Christian Richter
Kevin P. Fleming
Steve Murphy
Joshua Colp
Matt O'Gorman
Olle Johansson
Luigi Rizzo
BJ Weschke
North Antara
Mark Spencer
Matthew Fredrickson
Tilghman Lesher