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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48332 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48324 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48258 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48252 | tilghman | 2006-12-04 19:34:34 -0600 (Mon, 04 Dec 2006) | 14 lines
Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines
If the recording in the database is too large, it will fail to retrieve with
an mmap error. Not too sure why this doesn't happen when we put it in the
database, also, but since that doesn't seem to be broken, I'm not going to fix
it (at least until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385)
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r48234 | file | 2006-12-04 13:16:31 -0500 (Mon, 04 Dec 2006) | 9 lines
Blocked revisions 48233 via svnmerge
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r48233 | file | 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines
If the generic bridge tells us not to retry, and we have a frame to spit out then break the bridge. Props to markit in #asterisk-bugs for bringing this up.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48235 f38db490-d61c-443f-a65b-d21fe96a405b
This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48216 f38db490-d61c-443f-a65b-d21fe96a405b
Something is wrong in the agi directory. Asking for autoconfig.h.
I have it disabled locally, but no reason to commit that change.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48208 f38db490-d61c-443f-a65b-d21fe96a405b
- 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48200 f38db490-d61c-443f-a65b-d21fe96a405b
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r48195 | russell | 2006-12-01 22:50:58 -0500 (Fri, 01 Dec 2006) | 3 lines
Backport the comment containing the warning regarding the limitations on the
usage of this function. It is thread safe, but not technically reentrant.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48196 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) | 12 lines
FreeBSD 6.1 does not include wget by default. However, it has fetch which will
work just fine for our purposes of downloading the sounds packages. So, check
for both wget and fetch and the configure script and use what was found to
download them. If neither one was found, and sound packages are selected that
must be downloaded, the install process will print out an informative error
message indicating the situation.
Also, fix a couple places where "make" was hard coded into some output messages
by replacing them with the $(MAKE) variable.
(issue #8451, initial patch by pabelanger, with additional modifications by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48191 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo