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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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.
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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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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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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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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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- 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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in both the append and non-append modes. Also, always truncating the partial
write makes the behavior of the function more consistent, where in any type of
write, no partial result is left in the buffer.
Thanks for the feedback, luigi
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and also add a more verbose comment explaining why it is only needed in the
case of appending to the string for any curious readers that come along in the
future.
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an extra allocation on a path where we have way too many already.
Unfortunately the AMI-over-HTTP requires multiple copies,
because we need to generate a header, then the raw output to
an intermediate buffer, then convert it to html/xml, and
finally copy everything into a malloc'ed buffer because
that's what the generic_http_callback interface expects.
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ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name()
I am unsure whether the truncation of the string in case of a failed
attempt should be done unconditionally. See the XXX mark.
Russel, ideas ?
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to simplify the body of the main loop of the accepting thread.
Rename purge_unused() to purge_events() so one knows what the
function does.
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+ use a wrapper around ast_carefulwrite(), used in two places,
to make life easier when we decide to use a different interface
to the socket.
+ put an ast_verbose() message on astman_append on a case that
should never happen now that we use a temporary file for
AMI-over-HTTP sessions
+ document and slightly simplify process_events() by removing
unnecessary parentheses.
+ in get_input(), use ast_wait_for_input() instead of poll().
We may want to move to a completely non-blocking
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causing the http threads to do busy waiting around the socket...
Fix the mistake, sorry for the inconvenience!
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