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r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines
Merged revisions 309541 via svnmerge from
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r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines
Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file.
Also, prepend a newline to traceback output so that the main error message is on it's own line.
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r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines
Merged revisions 309494 via svnmerge from
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r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines
remove mysterious lua_pushvalue() that is never used
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(closes issue #16024)
Reported by: mnicholson
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Asterisk).
In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants.
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r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines
Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132.
(closes issue #18671)
Reported by: Igels
Patches:
pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
Tested by: Igels
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r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
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r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines
Merged revisions 309356 via svnmerge from
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r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines
Merged revisions 309355 via svnmerge from
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r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines
fix small memory leak
fix small memory leak caused by a string allocation that wasn't freed
(closes issue #18907)
Reported by: andy11
Patches:
asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
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(closes issue #18226)
Reported by: clegall_proformatique
Patches:
asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139)
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r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
Merged revisions 309255 via svnmerge from
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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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r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines
Fix consistency of CRLFs on HTTP headers that get sent out.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
18186-httpheadernewline.diff uploaded by qwell (license 4)
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r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines
Document CHANNEL(keypad_digits) and CHANNEL(no_media_path).
* Added XML documentation for CHANNEL(keypad_digits) and
CHANNEL(no_media_path).
* Tweaked XML documentation for CHANNEL(reversecharge).
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r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines
Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal.
Looks like an unintended change when sig_analog.c was extracted from
chan_dahdi.c.
Removed useless conditional around needed code and fixed resulting
compiler warning.
(closes issue #18667)
Reported by: enegaard
Patches:
issue18667.patch uploaded by enegaard (license 1197)
Tested by: enegaard
JIRA SWP-2965
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r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines
Merged revisions 309083 via svnmerge from
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r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines
Fixes thread blocking issue in the sip TCP/TLS implementation.
(closes issue #18497)
Reported by: vois
Patches:
issues_18497.diff uploaded by dvossel (license 671)
Tested by: vois, rossbeer, kowalma, Freddi_Fonet
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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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r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines
Merged revisions 308990 via svnmerge from
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r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines
Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled.
(closes issue #18815)
Reported by: irroot
Patches:
func_odbc.insert_nodata.patch uploaded by irroot (license 52)
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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
Merged revisions 308814 via svnmerge from
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r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
Merged revisions 308813 via svnmerge from
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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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r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines
Merged revisions 308722 via svnmerge from
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r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Merged revisions 308721 via svnmerge from
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r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines
silence gcc 4.2 compiler warning
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r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines
Merged revisions 308678 via svnmerge from
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r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines
Use remotesecret to authenticate with a remote party
The remotesecret option was only being used for outbound registration
and not for placing calls. This patch uses remotesecret on outbound
calls if it is set, otherwise secret is still used.
Review: https://reviewboard.asterisk.org/r/1107/
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r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines
sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.
(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
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audio ConfBridge, and other stuff
-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/
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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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r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines
Merged revisions 308414 via svnmerge from
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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.
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The callback handle_statechange (pbx.c) fails to release its data
pointer, leaking memory in the process.
Reported by: tzafrir
Patches:
18735_pbx_free_callback.diff uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/1110/
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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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r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines
A few more (copies of) files to ignore in this directory.
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r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines
added g729onlyA option for announce only AnnexA g.729 codec in
h.323 capabilities. Option can be global or per user/peer.
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r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines
Fix FreeBSD builds.
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r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines
ifdef __linux__ keepalive variables also
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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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don't have these options on sockets.
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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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r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line
Don't crash when forcing caller id.
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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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r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines
Merged revisions 307836 via svnmerge from
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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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r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines
Merged revisions 307792 via svnmerge from
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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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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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change malloc to ast_calloc again
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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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r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines
Fix a gaffe in the CCSS sample configuration.
Discovered by Philippe Lindheimer and pointed out on #asterisk-dev
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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.
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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.
(closes issue #18542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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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/
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