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r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) | 1 line
Add missing set of name valid flag when dialing.
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r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) | 14 lines
chan_local: resolve a deadlock.
This patch resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic realtime
extensions, which pulls autoservice into the picture when doing a dialplan
lookup.
(closes issue #18818)
Reported by: nic
Patches:
issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2)
Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik
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r315394 | pabelanger | 2011-04-25 22:18:50 -0400 (Mon, 25 Apr 2011) | 14 lines
Merged revisions 315393 via svnmerge from
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r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr 2011) | 7 lines
Add back CLI command 'dialplan save'
(closes issue #19140)
Reported by: lmadsen
Patches:
__20110419_dialplan_save.patch.txt uploaded by lmadsen (license 10)
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r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 Apr 2011) | 9 lines
When using MGCP realtime gateway definitions, random crashes occur.
Fixed incorrect linked list node removal for realtime gateways.
(closes issue #18291)
Reported by: nahuelgreco
Patches:
dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162)
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r315259 | russell | 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) | 24 lines
Merged revisions 315258 via svnmerge from
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r315258 | russell | 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) | 17 lines
Merged revisions 315257 via svnmerge from
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r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines
Be more flexible with unknown chunks in wav files.
This patch makes format_wav ignore unknown chunks instead of erroring
out on them.
(closes issue #18306)
Reported by: jhirsch
Patches:
wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156)
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r315213 | russell | 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) | 14 lines
Merged revisions 315212 via svnmerge from
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r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) | 7 lines
Don't link non-cached realtime peers into the peers_by_ip container.
(closes issue #18924)
Reported by: wdoekes
Patches:
issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717)
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r315053 | alecdavis | 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) | 23 lines
Merged revisions 315052 via svnmerge from
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r315052 | alecdavis | 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) | 16 lines
Merged revisions 315051 via svnmerge from
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r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) | 11 lines
chan_local:check_bridge() misplaced misplaced ast_mutex_unlock
if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked.
(closes issue #19176)
Reported by: alecdavis
Patches:
bug19176.diff.txt uploaded by alecdavis (license 585)
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r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) | 12 lines
chan_dahdi: Can't return to normal ring after distinctive ring on FXS
clear a previous distinctivering pattern before each new call
(closes issue #18985)
Reported by: bromont
Patches:
bug18985.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, bromont
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r314959 | mnicholson | 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) | 24 lines
Merged revisions 314958 via svnmerge from
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r314958 | mnicholson | 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) | 17 lines
Merged revisions 311203,314908 via svnmerge from
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r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines
Don't hold the pvt lock while streaming a file.
ABE-2756
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r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines
Prevent the login thread and the app threads from using the asterisk channel at the same time.
ABE-2756
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r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | 2 lines
Fix a few typos (shown by Lintian)
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r314780 | russell | 2011-04-22 09:02:23 -0500 (Fri, 22 Apr 2011) | 18 lines
Merged revisions 314778 via svnmerge from
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r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or
GET VARIABLE FULL. The bug report shows func_read returning garbage in
the result. It assumed that the buffer passed in was initialized, like many
other functions do. In the more common code path (through the dialplan), it
is initialized, so just initialize it here too.
(closes issue #19050)
Reported by: johnz
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PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI
spans. It is similar to the CLI command "pri show spans".
(closes issue #15980)
Reported by: dwery
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r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) | 1 line
Correct DAHDIShowChannels XML documentation.
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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Includes a new highly optimized and customizable
ConfBridge application capable of mixing audio at
sample rates ranging from 8khz-192khz.
Review: https://reviewboard.asterisk.org/r/1147/
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r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines
Merged revisions 314549 via svnmerge from
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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Review: https://reviewboard.asterisk.org/r/1157/
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Previously, the DAHDI format bit fields matched up with the Asterisk
bitfields. Since the Asterisk codec bit fields were replaced in r306010,
codec_dahdi needs to contain the formats itself. In the future, the DAHDI
formats should either change to something other than bitfields, or the
bitfields need to move from include/dahdi/kernel.h to
include/dahdi/user.h.
Signed-off-by: Shaun Ruffell <sruffell@digium.com>
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r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 Apr 2011) | 1 line
AST_CONTROL_XXX comment changes.
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not allwaya accurate.
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r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) | 4 lines
Initialize track pointer
ast_reentrancy_init checks to see if it is NULL before initializing with calloc
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r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) | 8 lines
Use SSLv23_client_method instead of old SSLv2 only.
(closes issue #19095)
(closes issue #19138)
Reported by: tzafrir
Patches:
no_ssl2.diff uploaded by tzafrir (license 46)
Tested by: russell, chazzam
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r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines
Merged revisions 314205 via svnmerge from
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r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines
Remove duplicate documentation from func_channel.c
(closes issue #18970)
Reported by: IgorG
Patches:
func_channel.c.doc.diff uploaded by IgorG (license 20)
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r314203 | lmadsen | 2011-04-19 09:24:25 -0500 (Tue, 19 Apr 2011) | 15 lines
Merged revisions 314202 via svnmerge from
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r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines
Update seconds to milliseconds in ast_verb output.
(closes issue #19084)
Reported by: smurfix
Patches:
app_dial.patch uploaded by smurfix (license 547)
Tested by: lmadsen, smurfix
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message to asterisk-dev)
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The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself. This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box. The controlling user number should be made configurable.
JIRA ABE-2738
JIRA SWP-2846
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r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines
The AsyncAGI command loop is lax in the value it returns for the return status.
* Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal
exits from the command loop such as hangup would return SUCCESS.
* The "asyncagi break" command now returns SUCCESS and is now the only way
to break the command loop with that status. Previously, it returned
FAILED.
* The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
is not sent. Previously, this happened because of an error setting up the
AGI pipes.
* All executed AGI commands now get an AsyncAGI Exec result event.
Previously, if the command returned failure (because of hangup), the
command loop just exited with FAILURE and did not send the AsyncAGI Exec
result event.
* Makes sure that the channel frame queue is empty on hangup.
Review: https://reviewboard.asterisk.org/r/1183/
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r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines
Unclear code in app_dial.c.
Make code formatting clear.
(closes issue #19134)
Reported by: oej
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r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
Remove the need for deadlock avoidance in chan_sip do_monitor.
Deadlock avoidance between the sip pvt and the pvt->owner is
very difficult. Now that channel's are ao2 objects, this complication
is no longer necessary. It turns out the pvt's msg queue only
exists because of deadlock avoidance (when deadlock avoidance fails
msgs were added to a queue to be processed later), so this goes away as well.
The technique used in the new sip_lock_pvt_full() function should
be used as a template for replacing all locations where deadlock
avoidance occurs between a channel tech_pvt and the pvt's owner.
My hope is that this will begin a reversal of the invalid channel
driver locking architecture we have been using for so long.
This patch also resolves an issue where the pvt->owner gets
unlocked during processing the msg queue.
(closes issue #18690)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/1182/
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r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
sip codec negotiation of dynamic rtp payloads error fix
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
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(closes issue #17907)
Reported by: wedhorn
Patches:
cleanup.stateringout.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
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r313860 | jrose | 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) | 17 lines
Merged revisions 313859 via svnmerge from
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r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | 10 lines
Fix a Tab Completion bug that occurs due to multiple matches on a substring.
Makes word_match function in cli.c repeat a search for a command string until
a proper match is found or the string is searched to the last point.
(closes issue #17494)
Reported by: ffossard
Review: https://reviewboard.asterisk.org/r/1180/
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This fixes a regression in the media architecture change
where video frames did not have their video mark set
correctly. dvossel wrote this. twilson kindly committed
this, mmichelson found the bug.
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r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines
Leftover debug messages unconditionally sent to the console.
Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
option enabled outputs the following debug messages unconditionally:
Dialing T1847555121 on 1
Dialing www2w on 1
* Made debug messages in my_dial_digits() normal debug messages that do
not get output unless enabled.
* Reworded some debug messages in my_dial_digits() to be clearer.
* Replace strncpy() with ast_copy_string() in my_dial_digits() which does
the same job better.
(closes issue #18847)
Reported by: vmikhelson
Tested by: rmudgett
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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.
There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation. The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities. A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.
The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.
For example, you may have a single button that when not lit, there is no
active CCSS request. When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel(). If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful. The actual request could ultimately fail. Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.
The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary. The idea is to allow some level of
customization as to the phone's behavior.
As an example, you may want the BLF key to go solid once you have
requested a callback. You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback. You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.
Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine. You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.
You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states. For example, you
may have an extension 3000 that is currently associated with device
SIP/3000. You could then create a feature code for that extension that
may look something like:
exten => *823000,hint,ccss:sip/3000
You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.
(closes issue #18788)
Reported by: p_lindheimer
Patches:
ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski
Review: https://reviewboard.asterisk.org/r/1105/
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r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) | 5 lines
Revert flushing stale AsyncAGI commands from -r313615.
It looks like it was intentional to leave any commands or in-flight
commands in the queue in case Async AGI is run again on the call.
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r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011) | 2 lines
Miscellaneous AGI diagnostic message cleanup and code optimization.
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r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) | 5 lines
* Add missing channel lock to handle_cli_agi_add_cmd().
* Flush any Async AGI commands left over from earlier Async AGI control of
the call.
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r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
Merged revisions 313579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
Merged revisions 313545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
Asterisk does not hangup a channel after endpoint hangs up.
If the call that the dialplan started an AGI script for is hungup while
the AGI script is in the middle of a command then the AGI script is not
notified of the hangup. There are many AGI Exec commands that this can
happen with. The reported applications have been: Background, Wait, Read,
and Dial. Also the AGI Get Data command.
* Don't wait on the Asterisk channel after it has hung up. The channel is
likely to never need servicing again.
* Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
in run_agi(). It previously only could return AGI_RESULT_SUCCESS or
AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
(closes issue #17954)
Reported by: mn3250
Patches:
issue17954_v1.8.patch uploaded by rmudgett (license 664)
issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
issue17954_v1.4.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA SWP-2171
(closes issue #18492)
Reported by: devmod
Tested by: rmudgett
JIRA SWP-2761
(closes issue #18935)
Reported by: nvitaly
Tested by: astmiv, rmudgett
JIRA SWP-3216
(closes issue #17393)
Reported by: siby
Tested by: rmudgett
JIRA SWP-2727
Review: https://reviewboard.asterisk.org/r/1165/
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(closes issue #19076)
Reported by: lmadsen
Patches:
__20110408-channel-description.txt uploaded by lmadsen (license 10)
Tested by: lmadsen
Review: https://reviewboard.asterisk.org/r/1163/
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r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines
Bring the dumpchan application inline with "core show channel".
* Added fields that are in "core show channel" to dumpchan output.
* Fixed reuse of formatbuf before the previous string stored there was
used by snprintf. All output strings now have their own buffer.
* Adjusted the buffer sizes to not be so abusive of the stack now that
there are more buffers.
Change requested by oej.
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IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
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also went ahead and fixed the problem it introduces before committing.
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r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line
fixing stupid mistake with putting code before variable declaration
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r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines
reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
stay in the dahdi_pvt structs for individual channels (causing them to just
continue adding the new ones to the list) and also there was a memory leak
causes by the conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process.
(closes issue #17450)
Reported by: nahuelgreco
Patches:
patch.diff uploaded by jrose (license 1225)
Tested by: tilghman, jrose
Review: https://reviewboard.asterisk.org/r/1170/
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r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines
Backport a restructuring change from trunk to make the next change stand out.
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r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines
Frames from the inbound channel should go to all outbound channels in app_dial.c.
In app_dial.c:wait_for_answer() frames from the inbound channel should be
sent to all outbound channels instead of only if there is just one
outbound channel.
Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of
the the outbound channels. This can happen if a blond transfer is done by
a remote switch on the inbound channel.
JIRA AST-443
JIRA SWP-2730
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r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines
Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output.
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r313279 | lmadsen | 2011-04-11 14:36:40 -0500 (Mon, 11 Apr 2011) | 21 lines
Merged revisions 313278 via svnmerge from
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r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines
Merged revisions 313277 via svnmerge from
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r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines
Fix detection of OpenSSL 1.0
(closes issue #19093)
Reported by: tzafrir
Patches:
detect_openssl_10.diff uploaded by tzafrir (license 46)
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