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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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is not reentrant, and the result of a previous call to the function is no
longer valid after calling it again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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")
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
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
........
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
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allow custom threadstorage init functions to return failure
use a custom init function for chan_sip's temp_pvt, to improve performance a bit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
use "attribute_XXX" instead of *__attribute__ ((XXX)) so we
can handle compiler/os dependencies in our compiler.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- add ast_asprintf() and ast_vasprintf()
- tweak doxygen comments
- simplify the definition of a flag macro
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Olle Johansson
Joshua Colp
Russell Bryant
Luigi Rizzo
Steve Murphy
Paul Cadach
TransNexus OSP Development
Matt O'Gorman
Kevin P. Fleming
Tilghman Lesher
Matthew Fredrickson
Jason Parker
Anthony LaMantia