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: https://origsvn.digium.com/svn/asterisk/trunk@48258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@48235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@48216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@48208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@48196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@48191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r48147 | file | 2006-11-30 13:19:55 -0500 (Thu, 30 Nov 2006) | 9 lines
Blocked revisions 48146 via svnmerge
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r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines
Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Olle Johansson
TransNexus OSP Development
Tilghman Lesher
Jason Parker
Joshua Colp
Russell Bryant
Kevin P. Fleming