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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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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
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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
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
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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.).
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- 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.)
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there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds.
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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happen off the first channel.
(closes issue #10867)
Reported by: phsultan
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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(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");
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and why this does not overwrite useful information.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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.
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where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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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).
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+ 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
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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
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(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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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(issue #9984, patches from eliel and dimas)
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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))
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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