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r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines
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r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines
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r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines
Make Asterisk less crashy.
Since we might not put a new translation path on the channel, go ahead and
set it to NULL right after destroying the old one to ensure we don't try
to free an invalid translation path later on.
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r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines
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r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines
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r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines
Fix false reporting of an error by set_format().
In the case that the native format was able to be changed to match the
new requested format, the code proceeded to attempt to build a translation
path, anyway. The result would be NULL, since no translation path is
necessary and resulted in this function thinking an error has occurred.
This case is now specifically caught and no attempt to build a translation
path is attempted.
Thanks to our automated tests and bamboo.asterisk.org for catching this problem
and making a whole lot of noise when things started failing. :-)
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r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
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r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
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r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
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r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
Merged revisions 295843 via svnmerge from
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r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
Merged revisions 295790 via svnmerge from
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r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.
Note that if the CLI command were done on the channel call leg associated
with B it works.
This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/).
The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.
The struct ast_channel._softhangup code is a mess. The variable is used
for several purposes that do not necessarily result in the call being hung
up. I have added doxygen comments to describe how the various _softhangup
bits are used. I have corrected all the places where the variable was
tested in a non-bit oriented manner.
The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.
JIRA SWP-2470
JIRA SWP-2489
(closes issue #18171)
Reported by: SantaFox
(closes issue #18185)
Reported by: kwemheuer
(closes issue #18211)
Reported by: zahir_koradia
(closes issue #18230)
Reported by: vmarrone
(closes issue #18299)
Reported by: mbrevda
(closes issue #18322)
Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/
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r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines
Merged revisions 295710 via svnmerge from
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r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
Fix cache of device state changes for multiple servers.
This patch addresses a regression where device states across multiple servers
were not being processing completely correctly. The code works to determine
the overall state by looking at the last known state of a device on each
server. However, there was a regression due to some invasive rewrites of how
the cache works that led to the cache only storing the last device state change
for a device, regardless of which server it was on.
The code is set up to cache device state change events by ensuring that each
event in the cache has a unique device name + entity ID (server ID). The code
that was responsible for comparing raw information elements (which EID is)
always returned a match due to a memcmp() with a length of 0.
There isn't much code to fix the actual bug. This patch also introduces a new
CLI command that was very useful for debugging this problem. The command
allows you to dump the contents of the event cache.
(closes issue #18284)
Reported by: klaus3000
Patches:
issue18284.rev1.txt uploaded by russell (license 2)
Tested by: russell, klaus3000
(closes issue #18280)
Reported by: klaus3000
Review: https://reviewboard.asterisk.org/r/1012/
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r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines
Analog lines do not transfer CONNECTED LINE or execute the interception macros.
Add connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code.
Note that if you create a three-way call in sig_analog before transferring
the call, the distinction of the caller/callee interception macros make
little sense. The interception macro writer needs to be prepared for
either caller/callee macro to be executed. The current implementation
swaps which caller/callee interception macro is executed after a three-way
call is created.
Review: https://reviewboard.asterisk.org/r/996/
JIRA ABE-2589
JIRA SWP-2372
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r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
Merged revisions 294277 via svnmerge from
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r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
Fix playback failure when using IAX with the timerfd module.
To fix this issue the alert pipe will now be used when the timerfd module is
in use. There appeared to be a race that was not solved by adding locking in the
timerfd module, but needed to be there anyway. The race was between the timer
being put in non-continuous mode in ast_read on the channel thread and the IAX
frame scheduler queuing a frame which would enable continuous mode before the
non-continuous mode event was read. This race for now is simply avoided.
(closes issue #18110)
Reported by: tpanton
Tested by: tpanton
I put tested by tpanton because it was tested on his hardware. Thanks for the
remote access to debug this issue!
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r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines
Avoid valgrind warnings for ast_rtp_instance_get_xxx_address
The documentation for ast_rtp_instance_get_(local/remote)_address stated that
they returned 0 for success and -1 on failure. Instead, they returned 0 if the
address structure passed in was already equivalent to the address instance
local/remote address or 1 otherwise. 90% of the calls to these functions
completely ignored the return address and passed in an uninitialized struct,
which would make valgrind complain even though the operation was technically
safe.
This patch fixes the documentation and converts the get_xxx_address functions
to void since all they really do is copy the address and cannot fail.
Additionally two new functions
(ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
times where the return value was actually checked. The
get_and_cmp_local_address function is currently unused, but exists for the sake
of symmetry.
The only functional change as a result of this change is that we will not do an
ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
API change, it shouldn't have a noticeable change in behavior.
Review: https://reviewboard.asterisk.org/r/995/
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r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines
Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call.
When a call is transfered by ECT or implicitly by disconnect in sig_pri or
implicitly by disconnect in chan_misdn, the connected line information is
not exchanged. The connected line interception macros also need to be
executed if defined.
The CALLER interception macro is executed for the held call.
The CALLEE interception macro is executed for the active/ringing call.
JIRA ABE-2589
JIRA SWP-2296
Patches:
abe_2589_c3bier.patch uploaded by rmudgett (license 664)
abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/958/
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r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) | 14 lines
Fixes recursive lock problem in manager.c
It is possible for a AMI session to freeze because of invalid
use of recursive locks during the EVENT processing. This
patch removes the unnecessary locks.
(closes issue #18167)
Reported by: sustav
Patches:
manager_locking_v1.diff uploaded by dvossel (license 671)
Tested by: sustav
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r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) | 9 lines
Resolve some compiler errors in ast_sockaddr_is_any().
These errors came up once this function was used from within netsock2.c.
The errors were like the following:
netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
The usage of a union here avoids this problem.
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r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines
Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
While testing chan_gtalk I noticed jabber was using my IPv6 address
and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
to return both IPv6 and IPv4 results. Adding a family parameter gives you
the ablility to choose.
Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
Review: https://reviewboard.asterisk.org/r/973/
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r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines
Merged revisions 291580 via svnmerge from
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r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines
Merged revisions 291577 via svnmerge from
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r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines
Don't ignore frames that have been queued when softhangup'd
When an outgoing call is answered and hung up by the far end *very* quickly, we
may not read any frames and therefor end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately
sets the _softhangup flag on the channel and then queues the HANGUP control
frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates
that a hangup request has been made (which it will if _softhangup is set). So,
we end up losing control frames.
This change makes __ast_read continue to read frames even if a soft hangup has
been requested. It queues a hangup frame to make sure that __ast_read() will
still eventually return NULL.
Much thanks to David Vossel for all of the reviews, discussion, and help!
(closes issue #16946)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/740/
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r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) | 16 lines
Fixes manager.c crash.
This issue was caused by improper use of the mansession lock and
manession_session lock. These two structures are confusing to begin
with so I'm not surprised this occurred. I fixed this by consistently
making sure we use each of these locks only to protect the data
in the corresponding structure. We had mismatched usage of these
locks which resulted in no mutual exclusivity occurring at all.
(closes issue #17994)
Reported by: vrban
Patches:
mansession_locking_fix.diff uploaded by dvossel (license 671)
Tested by: vrban
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r291075 | rmudgett | 2010-10-11 11:42:54 -0500 (Mon, 11 Oct 2010) | 22 lines
Merged revisions 291073 via svnmerge from
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r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines
Fixed infinite loop in verbose/debug message output.
Setting the module/filename specific message level and then changing it
resulted in the linked list being looped on itself. Traversing this
linked list is an infinite loop if what you are looking for is not in the
list.
Also plugged some CLI parsing holes in the associated CLI command:
* Removing a nonexistent module from the list actually added it with a
level of zero.
* Setting the non-module specific level to zero is now equivalent to
setting it to "off" as documented.
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r290864 | jpeeler | 2010-10-07 21:56:24 -0500 (Thu, 07 Oct 2010) | 23 lines
Merged revisions 290863 via svnmerge from
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r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290862 via svnmerge from
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r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines
Ensure editline cleanup occurs when Ctrl-C is pressed at control console.
A recent change was made to avoid a race condition on shutdown which only called
the end functions from the console thread. However, when pressing Ctrl-C the
quit handler is called from the signal handler thread.
(closes issue #17698)
Reported by: jmls
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r290255 | tilghman | 2010-10-04 18:23:11 -0500 (Mon, 04 Oct 2010) | 18 lines
Merged revisions 290254 via svnmerge from
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r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines
Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines
Merged revisions 289798 via svnmerge from
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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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r289340 | qwell | 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) | 22 lines
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r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines
Merged revisions 289338 via svnmerge from
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r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines
Allow a manager originate to succeed on forwarded devices.
The timeout to wait for an answer was being set to 0 when a device forwarded to another
extension. We don't always need the timeout set like this, so make it an optional
parameter, and don't use it in this case.
ABE-2544
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r289099 | bbryant | 2010-09-28 14:18:02 -0400 (Tue, 28 Sep 2010) | 28 lines
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r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines
Merged revisions 289094 via svnmerge from
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r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines
Fixes an issue with the Newchannel AMI event during the Masquerading process.
Fixes an issue with the Newchannel AMI event during the Masquerading process,
where no Newchannel AMI event was generated for the psuedo channel used during
the masquerading process.
(closes issue #17987)
Reported by: RadicAlish
Patches:
newchannel.patch.txt uploaded by RadicAlish (license 1122)
Tested by: RadicAlish
Review: https://reviewboard.asterisk.org/r/937/
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r288341 | russell | 2010-09-22 11:45:18 -0500 (Wed, 22 Sep 2010) | 25 lines
Merged revisions 288340 via svnmerge from
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r288340 | russell | 2010-09-22 11:44:13 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288339 via svnmerge from
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r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines
Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.
The handling of -c and console=yes should be the same, but they were not.
When you specify -c, it sets both a flag for console module and for asterisk
not to fork() off into the background. The handling of console=yes only set
console mode, so you would end up with a background process() trying to run
the Asterisk console and freaking out since it didn't have anything to read
input from.
Thanks to beagles for reporting and helping debug the problem!
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r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
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r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
Simplify locking code for REDIRECTING interception macro when forwarding a call.
Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep 2010) | 18 lines
ast_channel_masquerade: Avoid recursive masquerades.
Check all 4 combinations of (original/clonechan) * (masq/masqr).
Initially original->masq and clonechan->masqr were only checked.
It's possible with multiple masq's planned - and not yet executed, that
the 'original' chan could already have another masq'd into it - thus original->masqr
would be set, that masqr would lost.
Likewise for the clonechan->masq.
(closes issue #16057;#17363)
Reported by: amorsen;davidw,alecdavis
Patches:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis
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r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep 2010) | 14 lines
ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels.
Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run.
End result would be corrupted memory, and the frequent report 'Bad Magic Number'.
(closes issue #17801,#17710)
Reported by: notthematrix
Patches:
Based on bug17801.diff1.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/928
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r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines
Merged revisions 286681 via svnmerge from
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) | 9 lines
Rate limit calls to fsync() to 1 per second after astdb updates.
Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
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r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
Merged revisions 282467 via svnmerge from
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r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
Merged revisions 282430 via svnmerge from
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r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially
changing, so send a SRCCHANGE so that RTP-based media streams can get
a new SSRC generated to reflect the change. Original patch by addix
(along with lots of testing--thanks!).
(closes issue #17007)
Reported by: addix
Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
srcchange.diff uploaded by twilson (license 396)
Tested by: addix, twilson
Review: https://reviewboard.asterisk.org/r/862/
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines
Merged revisions 281912 via svnmerge from
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r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines
Merged revisions 281911 via svnmerge from
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r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged,
which is what used to happen. It seems that fixes were made to attempt limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change.
There are two other control frames sent in ast_channel_bridge that probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave
this change up to the discretion of resolving issue #17007.
For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540
(closes issue #17404)
Reported by: sdolloff
Patches:
bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff
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r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines
Merged revisions 281574 via svnmerge from
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r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines
Don't move the time threshold for running scheduled events on every iteration.
Instead, only calculate the time threshold each time ast_sched_runq() is called.
(closes issue #17742)
Reported by: schmidts
Patches:
sched.c.patch uploaded by schmidts (license 1077)
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Russell Bryant
Olle Johansson
Richard Mudgett
Jeff Peeler
Terry Wilson
Paul Belanger
Mark Michelson
Tilghman Lesher
David Vossel
Jason Parker
Matthew Nicholson
Brett Bryant
Alec L Davis
Tzafrir Cohen
