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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48302 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48277 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48258 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48188 f38db490-d61c-443f-a65b-d21fe96a405b
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48164 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, 29 Nov 2006) | 10 lines
Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)
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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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no output.
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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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Should really define a macro IS_AMI(s) so it is clear what
we want to do.
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and document it.
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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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opening an SSL socket.
We are almost ready to make this code available to other modules.
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Because https is more secure than http, it usually
makes sense to keep this service more open than the
one on the unencrypted port.
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function doing work on the socket. This is another generalization
to provide a generic mechanism to open TCP/TLS socket with a thread
managing the accpet and children threads managing the individual
sessions.
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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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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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probably 1.4/1.2 issue as well if someone can look into that.
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with the PLC code
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- 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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2 lines
Video will never reach Packet2Packet bridging and can do more harm then good.
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using the 'maxfiles' configuration option. (issue #7499 reported by rkarlsba)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2 lines
If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47860 | file | 2006-11-20 14:51:36 -0500 (Mon, 20 Nov 2006) | 10 lines
Merged revisions 47859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines
Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2 lines
Only remove/destroy the RTCP I/O item if it exists.
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document a bit more when manager events are delivered
to the clients.
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no need to bring this in 1.4, it is just code cleanup
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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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