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70 Commits (25cea89d90efca01c69f5749e4ee01b38b0c96b1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
2f24689b49 |
Merged revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
b2d959c7fa |
fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2a53f2ec98 |
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5a5cb18f54 |
Make sure we handle a uint32_t payload in ast_frdup()
(closes issue #14080) Reported by: fnordian Patches: frame.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1e8e12efcf |
Janitor, use ARRAY_LEN() when possible.
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3d4c0cd421 |
Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3a9c27459e |
Merged revisions 158072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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790fde68d9 |
Another batch of files from RSW. The remaining apps and a few more
files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d185405755 |
Janitor project to convert sizeof to ARRAY_LEN macro.
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f1e9371da8 |
- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bb386c84e7 |
Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ae52cd4a76 |
Merged revisions 114207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines It was possible for a reference to a frame which was part of a freed DSP to still be referenced, leading to memory corruption and eventual crashes. This code change ensures that the dsp is freed when we are finished with the frame. This change is very similar to a change Russell made with translators back a month or so ago. (closes issue #11999) Reported by: destiny6628 Patches: 11999.patch uploaded by putnopvut (license 60) Tested by: destiny6628, victoryure ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8f2ae67a3e |
But we can change the API here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0271088279 |
Merged revisions 111280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line Put this flag back so we don't change the API. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f59c496a81 |
Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8718878490 |
Merged revisions 106552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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cfc1df4c1a |
Whitespace changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e54da94808 |
Add a non-invasive API for application level manipulation of T38 on a channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it.
(closes issue #11873) Reported by: dimas Patches: v4-t38-api.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103799 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6c3a7a3e51 |
Add some missing control frames.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1807acb9b0 |
Merged revisions 99081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines Revert adding the packed attribute, as it really doesn't make sense why that would do any good. Fix the real bug, which is to do the check to see if the frame came from a translator at the beginning of ast_frame_free(), instead of at the end. This ensures that it always gets checked, even if none of the parts of the frame are malloc'd, and also ensures that we aren't looking at free'd memory in the case that it is a malloc'd frame. (closes issue #11792, reported by explidous, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2a91da6613 |
Merged revisions 99004 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines Have IAX2 optimize the codec translation path just like chan_sip does it. If the caller's codec is in our codec list, move it to the top to avoid transcoding. (closes issue #10500) Reported by: stevedavies Patches: iax-prefer-current-codec.patch uploaded by stevedavies (license 184) iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184) Tested by: stevedavies, pj, sheldonh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4fb04cb58a |
Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9387f036d8 |
- Fix the last set of places where incorrect assumptions were made about the
sample length with g722. It is _2_ samples per byte, not 1. This was all over the place, and I believed it, and it is what caused me to take so long to figure out what was broken. - Update copyright information on codec_g722. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e0ff5fef5c |
remove a bunch of useless #include "options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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915b97d300 |
move internal function declarations to include/asterisk/_private.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9335ace850 |
another bunch of include removals (errno.h and asterisk/logger.h)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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edc78d6023 |
improve linked-list macros in two ways:
- the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7c56918262 |
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ebe4050128 |
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b0f3e6097e |
Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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901522a116 |
Add packetization data for G.722.
(closes issue #10900) Reported by: andrew Patches: frame.diff uploaded by andrew (license 240) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d78463be1e |
Corydon posted this janitor project to the bug tracker and mvanbaak provided
a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string (closes issue #10843) Reported by: Corydon76 Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c7a9ec1691 |
(issue #10724)
Reported by: eliel Patches: res_features.c.patch uploaded by eliel (license 64) res_agi.c.patch uploaded by seanbright (license 71) res_musiconhold.c.patch uploaded by seanbright (license 71) pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded by moy (license 222) frame.c.patch uploaded by moy (license 222) manager.c.patch uploaded by moy (license 222) http.c.patch uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy (license 222) res_realtime.c.patch uploaded by eliel (license 64) res_odbc.c.patch uploaded by seanbright (license 71) res_jabber.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_agent.c.patch uploaded by eliel (license 64) chan_alsa.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71) Convert many CLI commands to the NEW_CLI format. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b4778de562 |
(closes issue #10225)
Reported by: klaus3000 Clean up AST_FORMAT_LIST list. It may have mattered in the old days to have undefined entries but these days it does not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a2b3357a9d |
Merged revisions 70360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun 2007) | 2 lines Put the speex packetization values back in but disable it when setting up the smoother. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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61544c5019 |
Merged revisions 70198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70198 | file | 2007-06-19 20:24:36 -0400 (Tue, 19 Jun 2007) | 2 lines Don't do packetization/smoother stuff with speex, it doesn't work. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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055d82cbce |
Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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897418858d |
corrected CLI 'core show codecs' syntax for issue 9945, thanks eserra.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9d05ff8ed5 |
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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21b53af31d |
Cosmetic changes. Make main source files better conform to coding guidelines and standards. (issue #8679 reported by johann8384)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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dcca8f345f |
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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68ff3c3575 |
Issue #8663 - Add passthrough support for MPEG4 (neutrino88).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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c5c17a1434 |
Merged revisions 49536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007) | 2 lines don't mark these allocations as 'cache' allocations when caching has been disabled ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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ec2ae45651 |
Merged revisions 49457,49460-49461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines make building of codec_gsm against the system GSM library actually work ........ r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines don't define this type either if LOW_MEMORY is enabled ........ r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines don't do frame header caching in the core if LOW_MEMORY is defined ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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9074555e37 |
- Add error handling to ast_parse_allow_disallow
- Use this in chan_sip configuration parsing git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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adca0ff14b |
Merged revisions 49006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) | 2 lines since these variables all have static duration, none of them need initializers (they default to zero anyway) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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d68c7c8ce6 |
Merged revisions 48987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) | 2 lines allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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d3bf06f272 |
Resolve some compiler warnings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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17a2888d2e |
Staticize one, and Constify a bunch of usage strings for CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |