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without a packet.
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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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(Issue #7610)
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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
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length, it is the initial length of the string when first allocated
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- 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.
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monitor a jabber connection over manager.
patches from 7673 and 7666 with minor changes.
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Thanks, sorry I didn't put this in forever ago.
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...field_set() macro, to optimize setting strings to empty, resetting strings to shorter contents, etc.) and to chan_iax2.c, to use stringfields in the user, peer, and pvt structs. Has been running stably on iaxtel, but while iaxtel has a large registration volume, it doesn't seem to have a high call volume. So far, it seems to reduce heap usage by over half. YMMV\! Please report any IAX bugs that might involve stringfields\!
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
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Small change to the fix in the report.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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- use appropriate types in some assignments
- use ast_strlen_zero()
- don't manually free cid fields since ast_set_callerid() will handle it
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r39056 | file | 2006-08-06 20:15:51 -0400 (Sun, 06 Aug 2006) | 2 lines
Reset our stream and vstream pointers back to NULL so that any generator that uses them (file based MOH) will not try to close them again. (issue #7668 reported by jmls)
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- don't include a newline character when building a response for an unknown
state
(issue #7645, jonty)
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variable
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38982 | russell | 2006-08-05 05:01:37 -0400 (Sat, 05 Aug 2006) | 6 lines
Always generate a Newstate event in ast_setstate() instead of making it a
Newchannel event if the state was AST_STATE_DOWN. The Newchannel event will
always be generated in ast_request(), so this just causes a duplicated
Newchannel event in some cases.
(issue #7506, repoted by capouch, fixed by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38972 | russell | 2006-08-05 04:08:48 -0400 (Sat, 05 Aug 2006) | 3 lines
remove duplicate queue log entry when the caller exits on a timeout
(issue #7616, ppyy)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38950 | russell | 2006-08-05 03:21:12 -0400 (Sat, 05 Aug 2006) | 3 lines
don't advertise that this function can set a SIP header when it can only
do reads
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38928 | russell | 2006-08-05 02:37:59 -0400 (Sat, 05 Aug 2006) | 3 lines
make sure the priv-callerintros directory exists before trying to create a file
there (issue #7659, patch by hads, with some modifications by me)
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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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list instead of getting the first entry and then removing it.
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and fixed by AuPix)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38825 | file | 2006-08-03 15:54:02 -0400 (Thu, 03 Aug 2006) | 2 lines
Treat the file as invalid if we have no valid formats for it (issue #7643 reported by KNK)
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hints by using func_odbc
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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
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r38761 | tilghman | 2006-08-03 00:22:19 -0500 (Thu, 03 Aug 2006) | 2 lines
Bug 7648 - Checking wrong count for plurality on new messages for Dutch language
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r38731 | kpfleming | 2006-08-02 14:29:56 -0500 (Wed, 02 Aug 2006) | 3 lines
fix brain-damage I introduced when trying to fix the CANCEL/BYE sending mechanism for pending INVITES
accept unknown 1xx responses as 183 responses (as RFC3261 mandates we should do)
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but more to come i guess.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38686 | kpfleming | 2006-08-01 18:07:06 -0500 (Tue, 01 Aug 2006) | 2 lines
ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)
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r38654 | file | 2006-08-01 15:20:05 -0400 (Tue, 01 Aug 2006) | 2 lines
Close the stream when file based MOH stop. This won't get rid of their position in the file but it will cause the translation path to be setup again. (issue #7634 reported by asimpson)
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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
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r38585 | file | 2006-07-31 13:09:10 -0400 (Mon, 31 Jul 2006) | 2 lines
Add missing code to bring transferee channel out of MOH/autoservice under certain circumstance (issue #7611 reported by guillecabeza with minor mods by myself)
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Essentially make sure an RTCP structure exists before trying to delete it's scheduled item. (issue #7514 reported by jmls fixed by AuPix)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38546 | russell | 2006-07-31 00:04:02 -0400 (Mon, 31 Jul 2006) | 2 lines
Make the frame counting done with TRACE_FRAMES defined thread-safe
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r38547 | russell | 2006-07-31 00:06:16 -0400 (Mon, 31 Jul 2006) | 2 lines
one more small tweak for thread-safety of TRACE_FRAMES
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Reformat the lines to break at column 80.
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