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These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
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Fix created in Huntsville together with Mark M (putnopvut)
(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej
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the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
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extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
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Also, remove some redundant logic I recently added in a fix.
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redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.
(closes issue #12513)
Reported by: mneuhauser
Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)
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temporarily.
(closes issue #11712)
Reported by: callguy
Patches:
11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham
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(Fix for AMI Originate)
(Closes issue #12007)
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up very quickly.
(issue #12515)
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Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!
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cases.
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ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)
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extension, then consider the extension on the channel before falling back
to the default.
(closes issue #12479)
Reported by: darren1713
Patches:
exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)
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These changes make sure that the reference count for sip_peer objects properly
reflects the fact that the peer is sitting in the scheduler for a scheduled
callback for qualifying peers or for expiring registrations. Without this, it
was possible for these callbacks to happen at the same time that the peer was
being destroyed. This was especially likely to happen with realtime peers, and
for people making use of the realtime prune CLI command.
(closes issue #9520)
Reported by: kryptolus
Committed patch by me
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other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
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redundant error
message.
Issue AST-15
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(closes issue #12476)
Reported by: davidw
Patch by me
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all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10)
(closes issue #12456)
Reported by: fnordian
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(Closes issue #12457)
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configured user don't override settings in general.
(closes issue #12458)
Reported by: tzafrir
Patches:
chanzap_users_sections.diff uploaded by tzafrir (license 46)
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telling me about this bug.
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If it is NULL, we can skip this channel, since it can't the one we're looking for.
(closes issue #9299)
Reported by: vazir
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capabilities.
(closes issue #12414)
Reported by: MVF
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the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen.
Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed.
(issue #12400)
Reported by: ztel
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(closes issue #11775)
Reported by: fujin
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(closes issue #12407)
Reported by: homesick
Patches:
uri_options-1.4.diff uploaded by homesick (license 91)
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during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.
(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann
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(for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor
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we should not send a BYE.
(closes issue #12392)
Reported by: fnordian
Patches:
chan_sip.patch uploaded by fnordian (license 110) with small modification from me
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7962/7965, 7975.
Thanks to Greg Oliver for providing me the required information.
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deadlocks. (Closes issue #12252)
Reported by: callguy
Patches:
20080319__bug12252.diff.txt uploaded by Corydon76 (license 14)
Tested by: callguy
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This fixes a for loop (in realtime_peer) to check all the ast_variables the loop was intending to test rather than just the first one. The change exposed the problem of calling memcpy on a NULL pointer, in this case the passed in sockaddr_in struct which is now checked.
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(closes issue #9077)
Reported by: pj
Patches:
earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn
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(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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(closes issue #10604)
Reported by: keepitcool
Patches:
branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@112766 f38db490-d61c-443f-a65b-d21fe96a405b
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makes the correct warning and debug messages print.
(closes issue #12361)
Reported by: one47
Patches:
chan_zap_deferred_digit.patch uploaded by one47 (license 23)
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early media, or if the channel has been answered.
(closes issue #11823)
Reported by: SDamm
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T38 SDP if need be.
(closes issue #11995)
Reported by: fall
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rolls over.
(closes issue #11923)
Reported by: mihai
Patches:
asterisk-fullvideo.patch uploaded by mihai (license 94)
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all the ast_variables the loop was intending to test rather than just the first one.
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to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.
So this is a revert of a revert...sort of.
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was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk
so that all scheduler functions are fixed at once.
I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.
(closes issue #12272)
Reported by: qq12345
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