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deprecated function)
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will show the version of each library respectively.
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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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basis in the register line. This comes from a Switchvox patch. (issue AST-24)
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r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines
Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)
(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej
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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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failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
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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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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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set with the channel pvt lock being held. #12512
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r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
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(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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(closes issue #12524)
Reported by: ctooley
Patches:
sip_qualify_peer.diff.2 uploaded by ctooley (license 136)
some modifications for trunk by Corydon76
Tested by: Corydon76
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r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines
Use consistent logic for checking to see if a call number has been chosen yet.
Also, remove some redundant logic I recently added in a fix.
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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(closes issue #12519)
Reported by: falves11
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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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r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines
Resolve a deadlock in chan_local by releasing the channel lock
temporarily.
(closes issue #11712)
Reported by: callguy
Patches:
11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham
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r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines
Ensure that when we set the accountcode, it actually shows up in the CDR.
(Fix for AMI Originate)
(Closes issue #12007)
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(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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r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines
Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
up very quickly.
(issue #12515)
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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines
Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!
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(closes issue #12514)
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r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines
Fix find_callno_locked() to actually return the callno locked in some more cases.
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r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines
Add 502 support for both directions, not only one... (see r114571)
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r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines
Treat a 502 just like a 503, when it comes to processing a response code
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r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines
When we receive a full frame that is supposed to contain our call number,
ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)
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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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phones that require it when sending them a special packet to get them to do something (such as reload their configuration).
(closes issue #9896)
Reported by: IgorG
Patches:
sipnotify-113980-v14.patch uploaded by IgorG (license 20)
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log_show_lock().
I gathered the code that filled the string, and put it in a different func which
I cryptically call "append_lock_information()".
Now, both log_show_lock(), and handle_show_locks() both call this code to do
the work. Tested, seems to work fine.
Also, log_show_lock was modified to use the ast_str stuff, along with checking
for successful ast_str creation, and freeing the ast_str obj when finished.
A break was inserted to terminate the search for the lock; we should never
see it twice.
An example usage in chan_sip.c was created as a comment, for instructional
purposes.
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines
Only drop audio if we receive it without a progress indication. We allow 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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#12425
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does not try to do some ADSI jazz.
(closes issue #12460)
Reported by: PerryB
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r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines
Clearing up error messages so they make a bit more sense. Also removing a redundant error
message.
Issue AST-15
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r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line
Only complete the SIP channel name once for 'sip show channel <channel>'
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astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
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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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r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines
Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug.
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(such as the i2050 softphone).
(closes issue #12398)
Reported by: c_hans
Patches:
chan_unistim_svn.diff uploaded by c (license 460)
Tested by: c_hans
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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines
The call_token on the pvt can occasionally be NULL, causing a crash.
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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return what the channel is bridged to instead of what they are *really* bridged to. Thanks Matt Nicholson!
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r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines
It is possible for the remote side to say they want T38 but not give any capabilities.
(closes issue #12414)
Reported by: MVF
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