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r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007) | 3 lines
ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug)
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r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) | 5 lines
Fix up a bunch of places where the iax2 pvt structure can disappear and the
code did not account for it and crashes.
(issues #9642, #9569, #9666, probably others ... based on the work by
stevedavies and mihai, with additional changes from me)
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r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) | 6 lines
Add comments for two functions that get called with the appropriate call locked,
but perform operations that could result in the pvt structure getting destroyed
before returning again, causing numerous seg faults all over the module.
(inspired by issues #9642, #9569, and #9666, and the work done by stevedavies
and mihai)
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over from my attempt at putting pvt structs in a hash table. It can cause
seg faults, and has no reason to stay.
(issue #9642, pointed out by stevedavies)
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r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) | 7 lines
Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that
was marked as the final transmission for a call, don't call iax2_destroy() for
that call while the global frame queue is still locked. There is a very nice
explanation of the deadlock in the report.
(issue #9663, thorough report and patch from stevedavies, additional positive
test reports from mihai and joff_oconnell)
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines
don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
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r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines
clear the 'delay PBX' flag when we are ready to start the PBX
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531, reported by and original patch by homesick, patch updated by me)
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"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
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probably others, too. I don't really have time to work on it at the moment,
so I am just going to revert it for now.
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r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines
When the IAX2 read callback gets called, return NULL instead of a "null frame".
This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286, john)
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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines
(This bug was reported to me by Kinsey Moore)
Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.
IAX2 media frames only carry the *source* call number. So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number. Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.
The way that chan_iax2 did this matching was extremely inefficient. It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it. It
would do this regardless of whether the call number was in use or not. So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations. (Note that the max call number is about 32k).
I have introduced a hash table of active IAX2 calls to improve this lookup
process. The hash is done on the IP address, port number, and call number.
So, for the previous example, a few lock/unlock operations may be done versus
25000 for each frame.
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines
Merged revisions 56406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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