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${ noResults }
125 Commits (b10671b0b8d4db2a83ebad32c76f797d30efba64)
Author | SHA1 | Message | Date |
---|---|---|---|
|
5fbd7ebd24 |
Merged revisions 79553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 lines (closes issue #10440) Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
22114b509d |
Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
5a1e2bfb50 |
Merged revisions 78172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 lines (closes issue #10355) Reported by: wdecarne Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f8483a0d04 |
Do a massive conversion for using the ast_verb() macro
(closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
77a75d46b2 |
Add a link to the list of assigned RTP payload types for convenience.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
5a96f8aa72 |
document how the RTP marker bit is passed for video frames,
and why this does not overwrite useful information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f1aadc8161 |
add an entry for h263plus in an empty slot of the rtp types.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8f4d728fe0 |
Merged revisions 77022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 lines set the sequence number in a frame for all frame types ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0e969271ae |
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5ac24b25d3 |
This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a7732f067 |
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c2603c1aeb |
resolve a compiler warning
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e950538bdd |
Small improvement to the STUN support so it can be used by
sockets other than RTP ones. The main change is a new API function in main/rtp.c (see there for a description) int ast_stun_request(int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer) which can be used to send an STUN request on a socket, and optionally wait for a reply and store the STUN_MAPPED_ADDRESS into the 'answer' argument (obviously, the version that waits for a reply is blocking, but this is no different from DNS resolutions). Internally there are minor modifications to let stun_handle_packet() be somewhat configurable on how to parse the body of responses. At the moment i am not committing any change to the clients, but adding STUN client support is extremely simple, e.g. chan_sip.c could do something like this: + add a variable to store the stun server address; static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add code to parse a config file of the form "stunaddr=my.stun.server.org:3478" (not shown for brevity); + right after binding the main sip socket, talk to the stun server to determine the externally visible address if (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock, &stunaddr, NULL, &externip); so now 'externip' is set with the externally visible address. so it is really trivial. Similarly ast_stun_request could be called when creating the RTP socket (possibly adding a struct sockaddr_in field in the struct ast_rtp to store the externalip). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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75e2b34c4d |
more cleanup, this time to stun_handle_packet(). Among other things:
+ mark a potentially dangerous write-past-end-of-buffer + localize some variables in the block generating stun replies. As before, not ready yet for a merge to 1.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74850 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3d41c1ce94 |
a little bit of code cleanup to rtp.c, mostly to function
ast_rtp_new_with_bindaddr(): 1. add comments to the logic of the main loop; 2. use a common exit point on failure so the cleanup is done only in one place; 3. handle failures in rtp_socket() in the main loop of the function; No functional changes except for #3 above, so it is not yet worthwhile merging this and other changes to 1.4 Once the cleanup work on this file will be complete (which among other things should include some extensions to the stun support) it might be a good thing to push all the changes to 1.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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deb98f98a0 |
add a bit of documentation on what the stun code in rtp.c does
(which is very little, at the moment). Eventually, when the functionality is extended, the changes can be merged back to 1.4. At the moment this is pointless. Note, this change is whitespace only. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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36b0448bc1 |
Merged revisions 72112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) | 3 lines Only output debug information related to RTCP timestamps when RTCP debug is turned on (issue #10066, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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792beb4686 |
Merged revisions 71915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4 lines Don't dereference a pointer that may be NULL here. Issue 10017. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71916 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5590f67f58 |
Convert so more logging to ast_debug (issue #10045, dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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80166c6de8 |
Conversions to ast_debug()
(issue #9984, patches from eliel and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6f98665672 |
Behold the magic of casting!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6a2af3c983 |
Merged revisions 71063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone. This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode)) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7a1c2d94bb |
Add manager events for RTCP statistics.
Also adds a new "reporting" permission for manager, since it can be incredibly spammy. This permission was discussed on the -dev mailing list some months back. Issue 8613, patch by johann8384, with some minor changes by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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80cdeaef55 |
Merged revisions 70727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 lines Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70729 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 |
|
9bec4f4b58 |
Merged revisions 70003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d0eaf1e389 |
Merged revisions 68922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, 12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines Bring RTP back to Asterisk at the end of a native bridge no matter what. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1d57ccb6f7 |
Fix a bunch of doxygen errors and document more things
(issue #9842, snuffy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68339 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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40df1fb464 |
Merged revisions 67650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, 06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b5089b4a58 |
Merged revisions 67071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | 2 lines Add a missing \n. (pointed out by jcmoore on IRC) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ed4726769a |
Merged revisions 66437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 lines Handle cases where a frame may have no data. (issue #9519 reported by dmb) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bcd2bd8294 |
Make this build on *my* machine again, and hopefully not break others.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e4191c375f |
Merged revisions 65863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines I like it when the RTP stack compiles myself... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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89b0e6049a |
Merged revisions 65842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines Fix the calculation of the RTT for RTCP. The previous code would result in oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65843 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b419fc1134 |
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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82d5673c81 |
Merged revisions 61707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines Avoid invalid seqno cycling detection. Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c21f118a65 |
Merged revisions 61697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines Remove a stray debug message introduced by a recent commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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16a080781d |
Merged revisions 61676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c4cd1b6761 |
Merged revisions 61674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines Issue #9554 - Improve RTCP (Dave Troy) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c4f42601d6 |
Merged revisions 59358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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08e3a9bdc8 |
Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ddca41798b |
Merged revisions 58783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2ab6ed30cd |
Merged revisions 58436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e7da006562 |
Merged revisions 58240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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aabe0abaee |
Merged revisions 57768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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75d387acbc |
Doxygen additions, corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ba32ee49d0 |
Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8f6d9918a7 |
Merged revisions 53434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 lines We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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dfb5ef7f55 |
Merged revisions 53429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines When parsing the NTP timestamp in a sender report message, you are supposed to take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |