https://origsvn.digium.com/svn/asterisk/branches/1.4
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r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines
Fix crash when moving audiohooks between channels.
Handle the scenario where we are called to move audiohooks between channels
and the source channel does not actually have any on it.
(closes issue #14734)
Reported by: corruptor
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r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines
Make chan_misdn BRI TE side normally defer channel selection to the NT side.
Channel allocation collisions are not handled by chan_misdn very well.
This patch simply avoids the problem for BRI only.
For PRI, allocation collisions are still possible but less likely since
there are simply more channels available and each end could use a different
allocation strategy.
misdn.conf options available:
te_choose_channel - Use to force the TE side to allocate channels.
method - Specify the channel allocation strategy.
(closes issue #13488)
Reported by: Christian_Pinedo
Patches:
isdn_lib.patch.txt uploaded by crich
Tested by: crich, siepkes, festr
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r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines
Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked.
(This is copied and pasted from the review request I made for this patch)
Asterisk has some odd behavior when queue weights are used. The current logic used when
potentially calling a queue member is:
If the member we are going to call is part of another queue and _that other queue has any
callers in it_ and has a higher weight than the queue we are calling from, then don't try
to contact that member. The issue here is what I have marked with underscores. If the
higher-weighted queue has any callers in it at all, then the queue member will be unreachable
from the lower-weighted queue. This has the potential to be really really bad if using a
queue strategy, such as leastrecent or fewestcalls, with the potential to call the same
member repeatedly.
The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works
well for this situation. With this set of changes, the logic used becomes:
If the member we are going to call is part of another queue, the other queue has a higher
weight than the queue we are calling from, and the higher weight queue has at least as many
callers as available members, then do not try to contact the queue member. If the higher
weighted queue has fewer callers than available members, then there is no reason to deny
the call to this member since the other queue can afford to spare a member.
Since the fix involved writing a generic function for determining the number of available
members in the queue, I also modified the is_our_turn function to make use of the new
num_available_members function to determine if it is our turn to try calling a member. There
is one small behavior change. Before writing this patch, if you had autofill disabled, then
if you were the head caller in a queue, you would automatically be told that it was your
turn to try calling a member. This did not take into account whether there were actually any
queue members available to take the call. Now we actually make sure there is at least one
member available to take the call if autofill is disabled.
(closes issue #13220)
Reported by: garychen
Review: http://reviewboard.digium.com/r/202/
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r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines
Backport state interface changes to app_queue from trunk.
After several issues raised on the Asterisk bugtracker against
the 1.4 branch were determined to be fixable with the state interface
change available in the 1.6.X series, it finally came time to just
suck it up and backport the change.
For a detailed explanation of what this change entails, the original
trunk commit for this feature may be found here:
http://svn.digium.com/view/asterisk?view=revision&revision=97203
In addition, the details for the use of this change to fix the problems
stated in issue #12970 may be found in the review request I made for
this change. It is linked below.
(closes issue #12970)
Reported by: edugs15
Review: http://reviewboard.digium.com/r/116
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r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
Improve our handling of T38 in the initial INVITE from a device.
We now answer with matching media streams to what is requested. If an INVITE
is received with both a T38 and RTP media stream this means we answer with both.
For any outgoing calls created as a result of this inbound one no T38 is requested
in the initial INVITE. Instead if we start receiving udptl packets we trigger a
reinvite on the outbound side.
(closes issue #12437)
Reported by: marsosa
Tested by: pinga-fogo, okrief, file, afu
Review: http://reviewboard.digium.com/r/208/
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The doxygen documentation has now been updated to state explicitly that I want
punctuation atthe end of the first sentence in a commit message. :).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a bridge to be created without actually being used.
In that scenario a timing file descriptor would be opened and not
closed. To fix this the timing file descriptor is now closed in the
destroy callback, not the thread function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AGI dialplan applications did not destroy the speech structure automatically
if it was not destroyed by the running AGI script. They will now do this.
(issue LUMENVOX-15)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was possible for no timer to become available between creating the bridge
and starting it. We now open a timer when creating it and keep it open until the
bridge is destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines
Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
If calls were placed using an IP address or hostname the global nat setting was copied over
but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
actions.
(closes issue #14546)
Reported by: acunningham
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When dvossel and I were doing some load testing last week, we noticed that we
could make Asterisk trunk lock up instantly when we started generating a bunch
of calls. The backtraces of locked threads were bizarre, and many were stuck
on an _unlock_ of an rwlock.
The changes are:
1) Fix a number of places where a backtrace would be loaded into an invalid
index of the backtrace array. It's an off by one error, which ends up
writing over the rwlock itself.
2) Ensure that in the array of held locks, we NULL out an index once it is
not being used so that it's not confusing when analyzing its contents.
3) Remove a bunch of logging referring to an rwlock operating being done
with "deep reentrancy". It is normal for _many_ threads to hold a
read lock on an rwlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines
pri loop TestClient/TestServer fails: server SEND DTMF 8
app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up.
(closes issue #12442)
Reported by: tzafrir
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This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we receive a T38 request negotiate control frame we should only attempt to do so
if the option is enabled on the dialog.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines
Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
When moving the cursor backward and pressing TAB to autocomplete, a NULL is put
in the line and we are loosing what we have already wrote after the actual
cursor position.
(closes issue #14373)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
Tested by: lmadsen
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r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines
Change NULL pointer check to be ast_strlen_zero.
The 'digit' variable is guaranteed to be non-NULL, so the if
statement could never evaluate true. Changing to ast_strlen_zero
makes the logic correct.
This was found while reviewing ast_channel_ao2 code review.
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The default codec configuration for chan_iax2 is bandwidth=low. I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
Fix a memory leak in res_monitor.c
The only way that this leak would occur is if Monitor were started
using the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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Russell Bryant
Joshua Colp
Richard Mudgett
Mark Michelson
Leif Madsen
Kevin P. Fleming
David Vossel
Eliel C. Sardanons
Tilghman Lesher