Age | Commit message (Collapse) | Author | Files | Lines |
|
f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.20-rc1@114929 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.20-rc1@114927 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.20-rc1@114924 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114891 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Fix created in Huntsville together with Mark M (putnopvut)
(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114890 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114880 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
local ones (inspired by r578 of asterisk-addons by tilghman)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114875 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.
(closes issue #12549)
Reported by: darren1713
Patches:
app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
(with modifications from me)
Tested by: putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114848 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
perfectly fine.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114829 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines
stop script from appending source code if run multiple times
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114823 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114708 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
may end up finding tds.h in /usr/local/include instead of /usr/include. If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114695 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12528)
Reported by: pukepail
Patches:
patch.diff uploaded by pukepail (license 431)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114689 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Also, remove some redundant logic I recently added in a fix.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114673 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
so that
the channel is not unlocked twice when using whisper mode.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114662 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12516)
Reported by: linuxmaniac
Patches:
diff_rev114611.patch uploaded by linuxmaniac (license 472)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114649 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114632 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
was being truncated two characters. This patch corrects the
problem.
(closes issue #12493)
Reported by: davidw
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114628 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
temporarily.
(closes issue #11712)
Reported by: callguy
Patches:
11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114624 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(Fix for AMI Originate)
(Closes issue #12007)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114621 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
up very quickly.
(issue #12515)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114608 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114603 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
would only work if the mansession_id cookie was first. Now, the code builds
a list of all of the cookies in the Cookie header. This fixes a problem
observed by users of the Asterisk GUI.
(closes AST-20)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114600 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Also, remove setting the amount of time to wait for a digit from 5 seconds back
down to 1/10 of a second. I believe this was so the beep didn't get played over
and over really fast, but a while back I put in another fix for that issue.
(closes issue #12498)
Reported by: jsmith
Patches:
app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114597 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #11575)
Reported by: sunder
Patches:
M11575_14_rev3.diff uploaded by junky (license 177)
bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114594 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
mansession_id cookie is coded to be limited to 8 characters of hex, and this
could break logins from 64-bit machines in some cases.
(inspired by AST-20)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114591 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
cases.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114587 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114584 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
large number that it can not print out a message informing the user and continue on.
(closes issue #12502)
Reported by: bcnit
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114579 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114571 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114558 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes
in this area that I have made in trunk/1.6.0, and things are happy again.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114550 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ast_write()
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114545 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #10890)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114542 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114537 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114522 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114322 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114299 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
"asterisk -rx 'help moh reload'" will hang. Reported via
-dev list, fixed by me.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114297 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114284 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114278 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114275 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
redundant error
message.
Issue AST-15
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114257 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12476)
Reported by: davidw
Patch by me
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114248 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114245 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
valid as an
input to SetCallingPres. (Closes issue #12472)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114242 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
accomplishes the same goal in a better way.
(issue #12470)
Reported By: atis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114230 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
variable defined
in the outer scope was never set properly, therefore making iterating through the channel
list always restart from the beginning. This bug would have affected anyone who called
chanspy without specifying a first argument.
(closes issue #12461)
Reported by: stever28
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114226 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
compiling before...
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114211 f38db490-d61c-443f-a65b-d21fe96a405b
|