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r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines
Allow dringXrange to properly default to 10, as was done in 1.4.
dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.
(closes issue #12536)
Reported by: bjm
Patches:
12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm
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r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines
Merged revisions 114891 via svnmerge from
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r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
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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r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines
Merged revisions 114880 via svnmerge from
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r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that 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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r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines
Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0.
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Merged to 1.6 because it fixes a crash.
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r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines
Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp.
(closes issue #12214)
Reported by: DEA
Patches:
chan_skinny-vm-events-v3.txt uploaded by DEA (license 3)
Tested by: DEA and me
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r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines
Merged revisions 114632 via svnmerge from
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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines
Re-invite RTP during a masquerade so that, for instance, an AMI
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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r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines
Merged revisions 51989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #12496)
Reported by: daniele
Patches:
misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele
Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.
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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line
added fix from #8899
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r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines
Merged revisions 114537 via svnmerge from
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r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines
If the dial string passed to the call channel callback does not indicate an
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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r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines
Merged revisions 114184 via svnmerge from
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r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines
use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to 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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r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines
Merged revisions 114083 via svnmerge from
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r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines
Several places in the code called find_callno() (which releases the lock on 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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r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines
Merged revisions 113927 via svnmerge from
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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines
We need to set the persistant_route [sic] parameter for the sip_pvt
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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r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines
Merged revisions 113784 via svnmerge from
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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines
If we receive an AUTHREQ from the remote server and we are unable to reply (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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r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines
Merged revisions 113681 via svnmerge from
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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines
If Asterisk receives a 488 on an INVITE (not a reinvite), then
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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r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) | 23 lines
Merged revisions 113013 via svnmerge from
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines
Merged revisions 113012 via svnmerge from
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(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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