https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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.
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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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Russell Bryant
Steve Murphy
Kevin P. Fleming
Joshua Colp
Luigi Rizzo
Olle Johansson
Jason Parker
Paul Cadach