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r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
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r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
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r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
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r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
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r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
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r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines
Fix Deadlock with attended transfer of SIP call
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
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r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data
from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer before
updating the caller party structure to see if the name or number was
changed.
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r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines
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r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
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r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
(closes issue #18168)
Reported by: FeyFre
Patches:
bug0018168.patch uploaded by FeyFre (license 1142)
Tested by: FeyFre, twilson
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-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
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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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/trunk@308372 f38db490-d61c-443f-a65b-d21fe96a405b
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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r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines
Merged revisions 308007 via svnmerge from
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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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List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting. Calls on hold or call-waiting
are not associated with any B channel.
JIRA LIBPRI-27
JIRA SWP-2547
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r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
No response sent for SIP CC subscribe/resubscribe request.
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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r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
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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r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines
Merged revisions 307535 via svnmerge from
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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
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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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The nativeformats field was being overwritten when it should have been
appended too. This caused some format capabilities to be lost briefly and
some log warnings to be output.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307433 f38db490-d61c-443f-a65b-d21fe96a405b
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.
(closes issue #18542)
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/trunk@307396 f38db490-d61c-443f-a65b-d21fe96a405b
From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.
(closes issue #17905)
Reported by: rcasas
Patches:
app_meetme.c.patch uploaded by rcasas (license 641)
Review: https://reviewboard.asterisk.org/r/874/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307359 f38db490-d61c-443f-a65b-d21fe96a405b
Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.
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r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines
Fix issue with verbose messages not showing on remote console.
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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qwell