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${ noResults }
154 Commits (4fb04cb58a15aafd7cf4be6d042bc0cccdbbefe9)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Russell Bryant
|
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 |
Joshua Colp
|
8e0dbcf7d7 |
Merged revisions 98325 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Olle Johansson
|
17afebc1a6 |
HUGE improvements to QoS/CoS handling by IgorG
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
18 years ago |
|
Joshua Colp
|
8a7064d3fc |
Merged revisions 92204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much. (closes issue #11483) Reported by: revolution Patches: rtp.diff uploaded by revolution (license 346) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
|
77ec19e255 |
Merged revisions 91637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) | 5 lines At the end of a call, when we're reporting, RTCP may already be partially torn down, so check for NULL dereference Reported by: blitzrage Patch by: tilghman (Closes issue #11450) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
985c9f5cfe |
Merged revisions 90588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
df7ba90b20 |
The following patch with updates for trunk. Works much better in trunk.
Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Luigi Rizzo
|
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 |
|
Luigi Rizzo
|
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 |
|
Luigi Rizzo
|
5663ff6518 |
fix breakage induced by previous mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Luigi Rizzo
|
5490960453 |
remove a bunch of duplicate includes
Reproduce with grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89348 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Luigi Rizzo
|
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 |
|
Tilghman Lesher
|
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 |
|
Joshua Colp
|
255e26c480 |
Drop the RTCP Read too short message to debug. There are some phones out there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jason Parker
|
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 |
|
Jason Parker
|
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 |
|
Jason Parker
|
65761cbd7a |
More changes to NEW_CLI.
Also fixes a few cli messages and some minor formatting. (closes issue #11001) Reported by: seanbright Patches: newcli.1.patch uploaded by seanbright (license 71) newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch uploaded by seanbright (license 71) newcli.5.patch uploaded by seanbright (license 71) newcli.6.patch uploaded by seanbright (license 71) newcli.7.patch uploaded by seanbright (license 71) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
a5122f03ad |
Merged revisions 85559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 lines Bring both DTMF begin and end frames up through to the core for DTMF feature handling. (closes issue #10826) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
c7ea8f9c87 |
Merged revisions 85552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 lines If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work. (closes issue #10943) Reported by: julianjm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
6f2e7b4310 |
Merged revisions 85057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4 lines Only update codec information if the channel has a technology private structure. (issue #10915) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
de64c85b54 |
Merged revisions 85023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4 lines Update codec information as well as address when doing hold reinvites. (issue #10868) Reported by: mavince ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
5b3347c715 |
Merged revisions 84818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 lines Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite. (closes issue #10868) Reported by: mavince ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
cfc8e90501 |
Merged revisions 84581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007) | 2 lines When an RFC 2833 event is sent that we don't recognize, ignore it, don't queue a NULL digit (closes issue #10877) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
fe1d4b1d04 |
Don't swap channel priority if using epoll as polling should/will only happen off the first channel.
(closes issue #10867) Reported by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
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 |
|
Russell Bryant
|
9f64905d4e |
Merged revisions 83432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, patch from qwell) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83433 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
44aacc96f1 |
Merged revisions 80974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4 lines (closes issue #10562) Reported by: idkpmiller Correct jitter value output in the CLI to be as expected. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
91d9c110af |
Merged revisions 80255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4 lines (closes issue #10526) Reported by: sinistermidget Revert commit from issue #10355 and return timestamp skew to 640. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
56b9568164 |
Don't reload a configuration file if nothing has changed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
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 |
|
Joshua Colp
|
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 |
|
Joshua Colp
|
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 |
|
Russell Bryant
|
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 |
|
Russell Bryant
|
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 |
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Luigi Rizzo
|
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 |
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Luigi Rizzo
|
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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Luigi Rizzo
|
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 |
Steve Murphy
|
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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Steve Murphy
|
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 |
|
Steve Murphy
|
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 |
|
Russell Bryant
|
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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Luigi Rizzo
|
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 |
|
Luigi Rizzo
|
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 |
|
Luigi Rizzo
|
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 |
|
Luigi Rizzo
|
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 |
|
Russell Bryant
|
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 |
|
Jason Parker
|
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 |
|
Russell Bryant
|
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 |
|
Russell Bryant
|
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 |
|
Joshua Colp
|
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 |