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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48580 f38db490-d61c-443f-a65b-d21fe96a405b
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Now the only module left using it is chan_sip.c
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for the verbose and logging messages so start with a
larger buffer from the beginning.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006) | 3 lines
remove some now-unnecessary explicit includes of autoconfig.h
clean up per-file dependencies during 'make clean'
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On passing remove presumably duplicate code to generate
the message for the manager_hooks:
in the previous version, the message was almost the same as the one sent
to regular sessions, with the exception of the empty line at the end, and
a few (presumably unintentional) differences e.g. timestamps,
debugging, and lowercase headers for "event" and "privilege".
now we reuse the same message as before.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006) | 2 lines
simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script
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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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This makes the code easier to follow and saves some
copies to intermediate buffers.
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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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r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2 lines
Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to.
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Not long ago i replaced lseek() with fseek() but
forgot that filr FILE's you need ftell to
give you the current position.
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equivalent to asprintf().
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place where it was used.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48476 f38db490-d61c-443f-a65b-d21fe96a405b
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2 lines
Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs.
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'Response: Success ' message even in case of
a recursive call.
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reducing indentation and normalizing loops.
While doing this, remove some unused variables,
fix an uninitialized string (idaction), and mark
some places where the behaviour is not what we would expect
(e.g. an empty context is reported as an error same as
a non-existent one). Given that this function is
not in 1.4, the above can be changed without too many
backward compatibility concerns.
Not applicable to 1.4 or below.
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matching word (e.g. "sip<TAB>"); this is implemented by this one-line change
- for (;; dst++, src += n) {
+ for (;src < argindex; dst++, src += n) {
However this code is not exactly trivial to understand, so
i am also adding some comments to help figuring out what it does.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48408 f38db490-d61c-443f-a65b-d21fe96a405b
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Long explaination:
The behaviour of the underlying malloc(0) differs depending on the
operating system. Some return NULL (SysV behaviour); some still
allocate a small chunk of memory and return a valid pointer (e.g.
traditional BSD); some (e.g. FreeBSD 6.x) return a non-null pointer
that causes a memory fault if used, even just for reading.
Given the above variety, better never call malloc(0).
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behaves differently in trunk for the case of a null prompt.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48388 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2 lines
Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one.
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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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47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri, 24 Nov 2006) | 6 lines
Doxygen update
- 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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r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri, 24 Nov 2006) | 1 line
This fix inspired by a patch supplied in bug 8189, which points out problems with the PLC code
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r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri, 24 Nov 2006) | 1 line
removed the svnmerge-integrated property from trunk; it's confusing svnmerge in newly created branches
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r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat, 25 Nov 2006) | 5 lines
set pointers to NULL after freeing memory to avoid multiple free()
probably 1.4/1.2 issue as well if someone can look into that.
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r48003 | oej | 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines
- Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't
have a clear understanding of the frame allocation/deallocation, so I just mark this
for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts...
- Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting
rtcp this way, but will need feedback from rtcp gurus. This should work for
video calls too, and possibly UDPTL.
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r48004 | oej | 2006-11-25 02:48:30 -0700 (Sat, 25 Nov 2006) | 2 lines
Changing ERROR to lesser level. Imported from 1.2/1.4
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r48008 | rizzo | 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines
generalize a bit the functions used to create an tcp socket
and then run a service on it.
The code in manager.c does essentially the same things,
so we will be able to reuse the code in here (probably
moving it to netsock.c or another appropriate library file).
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r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat, 25 Nov 2006) | 1 line
Updates to show linkset command
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r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat, 25 Nov 2006) | 2 lines
Add ss7 show linkset command
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r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat, 25 Nov 2006) | 1 line
Make sure we don't send a group reset on a group larger than 32 CICs
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r48012 | mattf | 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line
bug fix
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r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat, 25 Nov 2006) | 1 line
Make compiler happier
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r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1 line
Little fix so we use the right message
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r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9 lines
Merged revisions 48015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 line
A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it.
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r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat, 25 Nov 2006) | 9 lines
Merged revisions 48017 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 line
might as well also document the raw values of the flag vars
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r48019 | russell | 2006-11-25 23:55:33 -0700 (Sat, 25 Nov 2006) | 6 lines
- Add some comments on thread storage with a brief explanation of what it is
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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the infrastructure exposed in http.c earlier today.
As a bonus, now we can restart the session on a different
port just reloading the module.
On passing, fix a bug in the handling of 'enabled' in the configuration
file - previously, a missing "enabled=" line in manager.conf meant
"whatever the state was before" instead of a specific value.
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as described in
http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html
In detail, this commit does the following:
b) change the function get_input() to use fread() instead of read()
to collect the data. One can still do the ast_wait_for_input() on
the original descriptor returned by accept().
c) change the function send_string() to work on the FILE *.
As a side effect, this change now really guarantees that
we don't spend more than "writetimeout" milliseconds on
each line sent.
d) modify the function action_command() so that it creates a
temporary file descriptor to be passed to ast_cli_command(),
and then read back the data from the temp file and write it
to the output with send_string(). The code is similar to
what is done in generic_http_callback() to support AMI-over-HTTP.
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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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to English.
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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to bug guidelines)
Thanks 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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- remove coef_in.h and coef_out.h that was only included as data definitions in fskmodem.c
If you understand spanish, please help us translate the comments in fskmodem.c. Thanks!
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006) | 2 lines
Double-unlock error (reported by blitzrage on IRC)
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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r48152 | file | 2006-11-30 13:47:40 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines
Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader)
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