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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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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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines
Merged revisions 291577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines
Merged revisions 289338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines
Merged revisions 289094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r288340 | russell | 2010-09-22 11:44:13 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
