previously set are erroneously still set (Bug 6701). After discussion,
it was determined this should only be changed in trunk.
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines
move extern declaration for this option to a header file where it belongs
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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I don't know when the bug was introduced, but with the typical usage
c->fin = FRAMECOUNT_INC(c->fin)
the frame counters stay to 0.
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http://bugs.digium.com/view.php?id=8602
(i am not sure if there is still missing cast in
front of the alloca() call - being a macro this is
probably triggered only when actually used).
Add function ast_str_reset() to reinitialize the
string to an empty string instead of playing with
the internal fields.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) | 2 lines
since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself)
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renaming them to ast_str ... and putting the
struct ast_threadstorage pointer into the struct ast_str.
This makes the code a lot more readable.
At this point we can use these routines also to
replace ast_build_string().
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While doing this, add a bit of documentation, and slightly
extend the functionality as follows:
+ a max_len of -1 means that we take whatever the current size
is, and never try to extend the buffer;
+ add support for alloca()-ted dynamic strings, which is very
useful for all cases where we do an ast_build_string() now.
Next step is to simplify the interface by using shorter names
(e.g. ast_str as a prefix) and removing the _thread variant
of the functions by saving the threadstorage reference into
the struct ast_str. This can be done by overloading the
'type' field.
Finally, I will do my best to remove the convoluted interface
that results from trying to support platforms without va_copy().
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman)
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are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
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and implement services over tcp and/or tcp-tls.
This commit is nothing more than moving structure definitions
(and documentation) from main/http.c to include/asterisk/http.h
(temporary location until we find a better place), and removing the
'static' qualifier from server_root() and server_start().
The name change (adding the ast_ prefix as a minimum, and then
possibly a more meaningful name) is postponed to future commits.
Does not apply to other versions of asterisk.
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r48273 dealt with the comments and such, this deals with the code itself.
(This couldn't have been easily done if it weren't for 48273 - thanks again for that merbanan)
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implementing T.140 support in RTP.
T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired.
It defines a realtime text chat, much like the old "talk" application
in Unix.
T.140 is character by character in real time. It's not
the same as our current MESSAGE format - that is more like IM, but
can be gatewayed to MESSAGE with a text "codec" if needed.
More patches will follow, as soon as we've separated this code from
the video capabilities functions in the videocaps branch.
Code by John Martin, Aupix (disclaimer on file)
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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- Add some more documentation for the ast_dynamic_str_............() function
to document the behavior of the function in the case of a partial write.
Also, document the return value and note that the function should never need
to be called directly.
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as well as what the motivation is for using it.
- Add a comment by the declaration of ast_inet_ntoa() noting that this function
is not reentrant, and the result of a previous call to the function is no
longer valid after calling it again.
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- Document cause codes
- Document a bit more on channel variables - global, predefined and local
- Fix some doxygen in channel.h. Adding one comment for two definitions does not
work. They won't be copied to each.
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without duplicating the macro or the code:
/*!
* In many cases we need to print singular or plural
* words depending on a count. This macro helps us e.g.
* printf("we have %d object%s", n, ESS(n));
*/
#define ESS(x) ((x) == 1 ? "" : "s")
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Now you can specify a cli command as
"console autoanswer [on|off]"
which means the on|off argument is optional, or
"console {mute|unmute}"
which means the mute|unmute argument is mandatory.
The blocks in [] or {} do not necessarily need to be at the
end of the string.
Completions for the variant parts are generated automatically.
This should significantly simplify the implementation of
the various handlers.
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the NEW_CLI macro now supports extra arguments (to deprecate other commands).
use this to implement unload and reload, and remove some unused functions.
usual completion fixes (as these function accept multiple arguments).
The summary is still a bit inconsistent.
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This change basically simplifies the interface of the
new-style handler removing almost all the tricks used in
the previous implementation to achieve backward compatibility
(which is still present and guaranteed.)
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WATCH OUT: this changes the binary interface (ABI) for modules,
so e.g. users of g729 codecs need a rebuilt module (but read below).
The new way to write CLI handlers is described in detail in cli.h,
and there are a few converted handlers in cli.c, look for NEW_CLI.
After converting a couple of commands i am convinced that
it is reasonably convenient to use, and it makes it easier to fix the
pending CLI issues.
On passing, note a bug with the current 'complete' architecture:
if a command is a prefix of multiple CLI entries, we miss some
of the possible options. As an example, "core set debug" can
continue with "channel" from one CLI entry, and "off" or "atleast"
from another one.
We address this problem in a separate commit
(when i have figured out a fix, that is).
ABI issues:
I asked Kevin if it was ok to make this change and he said yes.
While it would have been possible to make the change without breaking
the module ABI, the code would have been more convoluted.
I am happy to restore the old ABI (while still being able
to use the "new style" handlers) if there is demand.
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debugging code in the manager.
At the moment the debugging code is very lightweight, if the option
is enabled manager messages also carry a sequence number and
the info where they have been generated e.g.
SequenceNumber: 10
File: chan_sip.c
Line: 11927
Func: handle_response_register
It is not worthwhile having this as a compile time option
right now, because the extra work involved at runtime is
just checking one variable.
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various PBX installations and checks if a module is loaded before using
it.
example IFMODULE(chan_sip3.so)
issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping
it updated.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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allow custom threadstorage init functions to return failure
use a custom init function for chan_sip's temp_pvt, to improve performance a bit
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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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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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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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use "attribute_XXX" instead of *__attribute__ ((XXX)) so we
can handle compiler/os dependencies in our compiler.h
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for uses in cases where you *know* that it will do no good. This patch was
inspired by file for use in some work of his on mixmonitor/chanspy.
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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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also, keep trying to dlclose() a module until it actually goes away, since it may have other modules it brought in when it was loaded (thanks PCadach for pointing this problem out to me)
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- add ast_asprintf() and ast_vasprintf()
- tweak doxygen comments
- simplify the definition of a flag macro
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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: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Olle Johansson
Tilghman Lesher
Kevin P. Fleming
Russell Bryant
Steve Murphy
Luigi Rizzo
Jason Parker
Paul Cadach
TransNexus OSP Development
Matt O'Gorman
Matthew Fredrickson
Anthony LaMantia