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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.4-rc2@308987 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.4-rc1@308636 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.4-rc1@308635 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.4-rc1@308633 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18874)
Reported by: cmaj
Patches:
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)
JIRA SWP-3172
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308622 f38db490-d61c-443f-a65b-d21fe96a405b
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Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
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r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines
Merged revisions 308413 via svnmerge from
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r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines
Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002
FAX-281
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enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308393 f38db490-d61c-443f-a65b-d21fe96a405b
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Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
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h.323 capabilities. Option can be global or per user/peer.
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r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines
Merged revisions 308002 via svnmerge from
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r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines
Fix regression that changed behavior of queues when ringing a queue member.
This reverts r298596, which was to fix a highly bizarre and contrived issue
with a queue member that called into his own queue being transferred back
into his own queue. I couldn't reproduce that issue in any way. I think one
of the other recent transfer fixes actually fixed this.
(closes issue #18747)
Reported by: vrban
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don't have these options on sockets.
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Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
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r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines
Need to retrieve the rows affected before using the associated variable.
(closes issue #18795)
Reported by: irroot
Patches:
20110211__issue18795.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines
Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once.
(issue #18156)
Reported by: asgaroth
Patches:
20110214__issue18156.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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A bug in AEL did not distinguish between the "s" extension generated by
AEL and an "s" extension that was required to exist by the chan_dahdi
(or another channel) that was not supplied with a starting extension.
Therefore, AEL made incorrect assumptions about what commands were
permissable in the context. This was fixed by making AEL generate a
different extension name. However, Dial and Queue make additional
assumptions about the name of the default gosub extension. Therefore,
they needed to be brought into line with a "macro" rendered by AEL (as
a gosub), without breaking traditional dialplans written without the
aid of AEL.
Related to (issue #18480)
Reported by: nivek
(closes issue #18729)
Reported by: kkm
Patches:
20110209__issue18729.diff.txt uploaded by tilghman (license 14)
018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
Tested by: kkm
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r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines
Merged revisions 307534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines
Remove color when executing commands via a remote console.
Essentially this makes '-x' imply '-n' on rasterisk. This was done in a
different and incomplete way previously, which I'm reverting here.
(issue #18776)
Reported by: alecdavis
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small fixes.
Interpret remote side H.225 version.
Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.
Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.
(issue 0018542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307509 f38db490-d61c-443f-a65b-d21fe96a405b
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Discovered by Philippe Lindheimer and pointed out on #asterisk-dev
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(closes issue #18776)
Reported by: alecdavis
Patches:
ast_deb_init.diff uploaded by lathama (license 1028)
Tested by: andrel, lathama
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307314 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18758)
Reported by: rgagnon
Patches:
branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
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r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines
Make sure to set parking dial context for non-default parking lots.
Since parking_con_dial isn't settable, set all parking lots to "park-dial".
(closes issue #17946)
Reported by: bluecrow76
Patches:
asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270)
modified by me
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(Reported by The_Boy_Wonder on IRC, fixed by me.)
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This code was reworked recently, and since the logchannel list hadn't been
created yet at this point, and it was a verbose message, it was being dropped
on the floor. Now it'll continue on to where it should be handled.
(closes issue #18580)
Reported by: pabelanger
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CC_EXTEN and CC_CONTEXT will allow you to determine the channel
and context that will be called when the recall occurs.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307065 f38db490-d61c-443f-a65b-d21fe96a405b
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Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage
(issue #16505)
Reported by: tzafrir
Patches:
asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir
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r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines
Merged revisions 306972 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines
Fix comparison for REFER Replaces tags with pedantic=yes
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r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines
Merged revisions 306965 via svnmerge from
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r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line
fix this line again
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r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines
Merged revisions 306960 via svnmerge from
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r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines
Backup file storing message duration is not used with IMAP_STORAGE, remove code.
The message duration is stored in the body of the email when using IMAP_STORAGE,
so nothing needs to happen with the backup file.
(closes issue #18718)
Reported by: kerframil
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r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines
Merged revisions 306864 via svnmerge from
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r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line
make this safer and fully correct, pointed out by Steve Davis
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More updates to the removed doc folder and
start updates to the man page.
(issue #16505)
Reported by: tzafrir
Tested by: lathama
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r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306672 via svnmerge from
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r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines
Don't try to pickup a call in the middle of a masquerade
If A calls B which doesn't answer and C & D both try to do a call pickup, it is
possible for ast_pickup_call to answer the call, then fail to masquerade one of
the calls because the other one is already in the process of masquerading. This
patch checks to see if the channel is in the process of masquerading before
call before selecting it for a pickup.
Review: https://reviewboard.asterisk.org/r/1094/
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r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306617 via svnmerge from
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r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines
Don't allow a REFER w/replaces to replace its own dialog
Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
header that matches the dialog of the REFER. This would be a situation like A
calls B, A calls C, A transfers B to A, which is just silly. This patch makes
the transfer fail instead of making Asterisk freak out and forget to hang other
channels up.
Review: https://reviewboard.asterisk.org/r/1093/
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By waiting to call the callback macro after the CC_INTERFACES,
extension, priority, and context have been set, this information
can be accessed more easily within the callback macro.
Reported by Philippe Lindheimer.
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r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines
Don't fallthrough to 'unknown' in the 'ringing' case.
This could cause improper exits from the queue.
(closes issue #18499)
Reported by: zaltar
Patches:
app_queue.patch uploaded by zaltar (license 1148)
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Each fork in the dial could be redirected and confuse the caller. For
ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN
redirects calls in sequence not in parallel.
* Also fixed a formatting inconsistency in app_dial.c and make a warning
message more useful about what frame type could not be written.
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Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
has caused locking problems. Both of these functions lock the channel when
the channel argument is passed in!
In this case, the suspected problem (the backtrace makes it impossible to tell)
was the private being locked in sip_set_rtp_peer and then:
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper
(Traced to verify that the fix was only required in 1.8 and later.)
(closes issue #18491)
Reported by: cmaj
Patches:
chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
Tested by: cmaj
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r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines
Merged revisions 306119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines
Set hangup cause in local_hangup
When a call involves a local channel (like SIP -> Local -> SIP), the hangup
cause was not being set. This resulted in SIP channels sometimes getting a
503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+
this also can cause issues with CCSS that involve a local channel. This patch
sets the hangupcause for one side of the local channel to the other in
local_hangup for outbound calls.
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r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines
Set exception on channel in parking thread when POLLPRI event detected.
This is done just to make the code be equivalent to the old select code. As
noted in 303106 the same issue was already fixed in this branch, but the
exception was not set on the channel in the case of POLLPRI. The reason that
this did not cause a problem here is because in 122923 the check in __ast_read
to check the exception flag was removed.
(related to #18637)
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(issue #18713)
Reported by: lathama
Patches:
snom_dir.diff uploaded by lathama (license 1028)
Tested by: lathama
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
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r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
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(closes issue #18731)
Reported by: marioabajo
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Adding links to http(s)://wiki.asterisk.org
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Adding links to http(s)://wiki.asterisk.org
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305798 f38db490-d61c-443f-a65b-d21fe96a405b
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