Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no
to enable dialplan faxdetect allowing more flexibility.
as soon as a fax tone is detected the framehook is removed.
there is a penalty involved in running this framehook on
non G711 channels as they will be transcoded.
CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice.
(Closes issue ASTERISK-18569)
Reported by: Myself
Reviewed by: Matthew Nicholson, Kevin Fleming
Review: https://reviewboard.asterisk.org/r/1116/
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r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
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r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
Forgot to svn add new files to r337595
Part of Generating security events for chan_sip
(issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
Reviewboard: https://reviewboard.asterisk.org/r/1362/
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r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
Make ast_pbx_run() not default to s@default if extension is not found
Review: https://reviewboard.asterisk.org/r/1446/
This is a bug - or architecture mistake - that has been in Asterisk for a
very long time. It was exposed by the AMI originate action and possibly
some other applications. Most channel drivers checks if an extension
exists BEFORE starting a pbx on an inbound call, so most calls will
not depend on this issue.
Thanks everyone involved in the review and on IRC and the mailing list
for a quick review and all the feedback.
(closes issue ASTERISK-18578)
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r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines
Change strictrtp option to default to yes in the RTP module
Suggested by Kapejod on Facebook
Review: https://reviewboard.asterisk.org/r/1448/
(closes issue ASTERISK-18587)
Thanks for quick feedback to kpfleming and Tilghman
--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M CHANGES
M configs/rtp.conf.sample
M res/res_rtp_asterisk.c
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r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
Curently only 80 bit is supported.
The outgoing invite will use the taglen of the incoming invite preventing
one-way audio.
(Closes issue ASTERISK-17895)
Review: https://reviewboard.asterisk.org/r/1173/
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r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make
it behave like a normal call bridge. When the second to last person exists, the last person
will be kicked out of the conference when this option is enabled.
(closes issue ASTERISK-18234)
Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden
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This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending
with congestion in a way that is unique from other unanswered calls.
(closes issue ASTERISK-14842)
Reported by: Alec Davis
Patches:
cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines
Merged revisions 332100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines
Fix multiple parking issues.
JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot.
JIRA ASTERISK-17870
Cannot retrieve parked calls.
JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call and does not.
JIRA AST-576
Issues with parking lots
* Removed searching for parking lots by extension. Parking lots can only
be found by the parking lot name since parking lot access extensions and
spaces are not guaranteed to be unique.
* Added parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications.
* Add parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access.
(closes issue ASTERISK-17183)
Reported by: David Cabrejos
Tested by: rmudgett, David Cabrejos
(closes issue ASTERISK-17870)
Reported by: Remi Quezada
(closes issue ASTERISK-17430)
Reported by: Philippe Lindheimer
JIRA ASTERISK-17452
Parking_offset not used
JIRA AST-624
'next' setting for findslot does nothing
* Reimplemented since findslot feature option broken by -r114655.
(closes issue ASTERISK-17452)
Reported by: David Woolley
Tested by: rmudgett
JIRA ASTERISK-15792
Dialplan continues execution after transfer to park.
This happens for DTMF attended transfer, DTMF blind transfer, and DTMF
one-touch-parking if the party initiating these features also initiated
the call.
* Fixed the return code from the affected builtin features when parking a
call.
(closes issue ASTERISK-15792)
Reported by: Mat Murdock
Tested by: rmudgett, twilson
JIRA AST-607
The courtesytone is not playing to the expected call when picking up a
parked call.
This is mostly a documentation problem. However, the option is not reset
to the default when features.conf is reloaded.
* Updated features.conf.sample documentation for courtesytone and
parkedplay options.
* Reset the parkedplay option to default when features.conf is reloaded.
JIRA AST-615
AMI Park action followed by features reload results in orphaned channels
in parking lot.
* Reloading features.conf will not touch parking lots that have calls
still parked in them. Reload again at a later time.
Misc additional fixes:
* Added unit test for parking lot dialplan usage checking.
* Made update connected line when a parked call is retrieved from a
parking lot.
* Made retrieved parked call stop ringing or MOH depending upon how the
call was waiting in the parking lot.
* Made CLI "features show" indicate if the parking lot is enabled for use.
* Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to
specify the parking lot access extension.
* Made AMI ParkedCalls action ParkedCall events have a Parkinglot header.
* Made AMI ParkedCalls action ParkedCallsComplete event have a Total
header.
* Fixed potential deadlock from AMI Park action holding channel locks
while calling masq_park_call().
* Fixed several places where ast_strdupa() were used inside of loops.
(Mostly fixed by refactoring the loop body into its own function.)
* Fixed copy_parkinglot() copying too much from the source parking lot.
Extracted the parking lot configuration settings into struct
parkinglot_cfg.
* Refactored courtesytone playing code to put the channel not playing the
tone in autoservice.
* Fix when pbx-parkingfailed is played that the other channel is put in
autoservice if it exists.
* Fixed parkinglot reference leak in parked_call_exec() error paths.
* Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to eliminate
redundancy.
* Fixed feature reload to better protect parking lots.
* Fixed parking lot container reference leak in handle_parkedcalls().
* Fixed the total count in handle_parkedcalls().
Review: https://reviewboard.asterisk.org/r/1358/
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r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
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In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1341/
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r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines
Bump the AMI protocol version to 1.2
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
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r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines
Replace AMI Unlink events with Bridge events
A previous update converted some of the Link and Unlink events to
Bridge events, but a couple of Unlink events were missed. This patch
rectifies the situation.
(closes issues ASTERISK-17455)
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There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session
(closes issue ASTERISK-16795)
Reported by: kobaz
Tested by: kobaz,loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.
Big thanks to irroot for porting this code to use the framehooks api.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
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r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
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r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines
Add ConnectedLineNum/Name headers to output of AMI action Status.
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
(closes issue #18158)
Reported by: gareth
Patches:
svn-292308.diff uploaded by gareth (license 208)
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