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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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probably 1.4/1.2 issue as well if someone can look into that.
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to ast_stream_and_wait() was redundant. Almost all calls just used chan->language, and seeing how chan is the first argument, this certainly seems redundant. A change of language could just as easily be done by simply changing the channel language before calling.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10 Окт 2006) | 1 line
CHANNEL() function sometime mix parameter and value
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2 lines
We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines
Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments"
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006) | 3 lines
Pass through a frame if we don't know what it is, rather than trying to pass a
NULL, which will segfault a channel driver (Bug 8149)
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be called for each thread specific object after they are allocated. Note that
there was already the ability to define a custom cleanup function. Also, if
the custom cleanup function is used, it *MUST* call free on the thread
specific object at the end. There is no way to have this magically done that
I can think of because the cleanup function registered with the pthread
implementation will only call the function back with a pointer to the
thread specific object, not the parent ast_threadstorage object.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) | 7 lines
Don't attempt to access private data members of the pthread_mutex_t object,
because this does not work on all linux systems. Instead, just access
the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is
enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
DEBUG_THREADS on as well.
(issue #8139, me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines
optimize the 'quick response' code a bit more... no more malloc() or memset() for each response
expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed
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#7840 reported by moy)
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patch provided in bugnote, with minor changes.
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number and CallerIDName for name. (issue #7976 reported by suhler)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines
Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006) | 2 lines
all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from...
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minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42600 | file | 2006-09-09 16:24:19 -0400 (Sat, 09 Sep 2006) | 2 lines
Only truly consider the channel in the same format if the format matches the raw format OR if a translation path already exists to translate between them. (issue #7887 reported by softins & issue #7803 reported by alvaro_palma_aste). Thanks goes to stubert for giving me access to a box and showing me a scenario where this occured.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42452 | file | 2006-09-08 14:50:43 -0400 (Fri, 08 Sep 2006) | 2 lines
Swap spies during masquerading
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spy are when it comes to spying on a channel. The core is responsible for adding a spy to a channel, feeding frames into the spy, removing the spy from the channel, and notifying the spy that is has been detached. The spy is responsible for reading frames in, and cleaning itself up. Each side will not try to do the other's job.
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very stupid thing to do. It ends up duplicating the frame twice, linking in
one of them and setting the tail pointer to the other one. Sorry ...
Thanks to file for pointing out the breakage!!! file rocks.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r41690 | file | 2006-09-01 12:28:08 -0400 (Fri, 01 Sep 2006) | 2 lines
Don't treat an unexpected control subclass as voice (issue #7858 reported by PCadach)
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ignore these.
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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translated frame to the spies, *not* the original frame. Thanks PCadach!
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40994 | russell | 2006-08-24 15:41:26 -0400 (Thu, 24 Aug 2006) | 11 lines
Fix a few issues related to the handling of channel variables
- in pbx_builtin_serialize_variables(), the variable list traversal would stop
on a variables with empty name/values, which is not appropriate
- When removing the GROUP variables, use AST_LIST_REMOVE_CURRENT instead of
AST_LIST_REMOVE
- During masquerading, when copying the variables list from one channel to the
other, using AST_LIST_INSERT_TAIL is not valid for appending a whole list.
It leaves the tail pointer of the list invalid. Introduce a new macro,
AST_LIST_APPEND_LIST that appends a list properly.
(issue #7802, softins)
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- free the string fields allocation if ast_create_channel() failes to open the
alert pipe
- formatting tweaks
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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)
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