Age | Commit message (Collapse) | Author |
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releasing video src.
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r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
Misc minor changes in chan_sip.
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
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Review: https://reviewboard.asterisk.org/r/1288/
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* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
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r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines
Remove extra 'the'.
Reported by Vlad Povorozniuc
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Review: https://reviewboard.asterisk.org/r/1265/
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r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines
Use correct syntax for 'sip notify snom-reboot'
(closes issue ASTERISK-17915)
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r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines
Also document the 'queue-minute' option.
(closes issue #19386)
Reported by: juanmol
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Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
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r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
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The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
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r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines
Documenting an observed behavior of features in features.conf. Since parkinglots use an
integer for the parkinglot extensions, leading zeros specified in the configuration file
are ignored.
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(closes issue #18462)
Reported by: joscas
Patches:
bug_18462.diff uploaded by snuffy (license 35)
cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180)
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r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines
Remove unused directory and clear up some documentation.
(closes issue #19193)
Reported by: bchia
Patches:
cel-csv.diff uploaded by lathama (license 1028)
Tested by: lathama, Marquis42
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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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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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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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(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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r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines
Merged revisions 312764 via svnmerge from
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r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines
Merged revisions 312761 via svnmerge from
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r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines
Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate.
AST-2011-005
(closes issue #18996)
Reported by: tzafrir
Tested by: mnicholson
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r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines
Incorrect default example; the field is actually internally named "clid", not "callerid".
(closes issue #19040)
Reported by: wcselby
Tested by: tilghman
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r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines
Merged revisions 311049 via svnmerge from
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r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines
Merged revisions 311048 via svnmerge from
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r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines
Remove extra quote in indications.conf
Picking low hanging fruit.
(closes issue #18971)
Reported by: IgorG
Patches:
based on indications.conf.sample.diff uploaded by IgorG (license 20)
Tested by: IgorG
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r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines
Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent.
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Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.
Review: https://reviewboard.asterisk.org/r/1134/
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(closes issue #16024)
Reported by: mnicholson
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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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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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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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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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Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.
The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.
JIRA SWP-2845
JIRA ABE-2736
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The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
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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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Adding links to http(s)://wiki.asterisk.org
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Adding links to http(s)://wiki.asterisk.org
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sip show settings reports qualifyfreq in milliseconds.
sip.conf configures qualifyfreg in seconds.
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r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines
Add alternative name for config option.
The SIP sample configuration had "tlscadir" as the option name, but chan_sip
used the more correct "tlscapath". Now both are accepted.
Discovered (sort of) by a user on IRC in #asterisk
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r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines
Add connected line chan_dahdi.conf pricpndialplan option.
* Added from_channel value to prilocaldialplan option.
JIRA ABE-2731
JIRA SWP-2842
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r304186 | seanbright | 2011-01-26 15:23:48 -0500 (Wed, 26 Jan 2011) | 16 lines
Merged revisions 304181 via svnmerge from
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r304181 | seanbright | 2011-01-26 15:22:47 -0500 (Wed, 26 Jan 2011) | 9 lines
Merged revisions 304159 via svnmerge from
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r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan 2011) | 1 line
Make sure the sample queues.conf is properly commented.
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r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
Merged revisions 303008 via svnmerge from
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r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
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r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
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r302417 | seanbright | 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) | 16 lines
Merged revisions 302416 via svnmerge from
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r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) | 9 lines
Remove references to priorityjumping from the sample extensions.conf.
Priority jumping was removed from pbx_config in r68970.
(closes issue #18622)
Reported by: kshumard
Patches:
extensions.conf.sample.patch uploaded by kshumard (license 92)
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r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines
Document "encryption" option in sip.conf.sample
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r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines
Merged revisions 301730 via svnmerge from
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r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines
Add static entry for split Polycom 332 firmware.
(closes issue #18607)
Reported by: cjacobsen
Patches:
polycom_331.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
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r301311 | pabelanger | 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) | 9 lines
Merged revisions 301310 via svnmerge from
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r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan 2011) | 2 lines
Fix a logic issue when passing context ARG
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r300433 | lmadsen | 2011-01-04 15:00:55 -0600 (Tue, 04 Jan 2011) | 15 lines
Merged revisions 300431 via svnmerge from
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r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) | 7 lines
Add some documentation to users.conf.sample.
(closes issue #18531)
Reported by: lathama
Patches:
users.conf.sample2.diff uploaded by lathama (license 1028)
Tested by: lathama
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hold.
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
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(closes issue #17979)
Reported by: tilghman
Patches:
20100911__for_blitzrage.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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r299312 | pabelanger | 2010-12-20 19:44:08 -0500 (Mon, 20 Dec 2010) | 8 lines
Correct typo with USER_DEFINED event.
(closes issue #18461)
Reported by: joscas
Patches:
cel.conf.sample.diff uploaded by lathama (license 1028)
Tested by: lathama, joscas
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r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) | 10 lines
Fix parsing of mwi => lines in sip.conf
Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(closes issue #18350)
Reported by: gbour
Tested by: Marquis, gbour
Review: https://reviewboard.asterisk.org/r/1053/
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