The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location. Previously, it was only
possible to redirect both channels to the same place.
(closes issue #15853)
Reported by: haakon
Patches:
trunk-manager.c.patch uploaded by haakon (license 880)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
conference.
* Page has a new option 'A(x)' which will playback an announcement
simultaneously to all paged phones (and optionally excluding the caller's one
using the new option 'n') before the call is bridged.
To add the new option to meetme, the conference flag options had to be extended
to 64 bits.
(closes issue #14365)
Reported by: dferrer
Patches:
page_announce.patch uploaded by dferrer (license 525)
modified by me
Review: https://reviewboard.asterisk.org/r/188/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X
versions. The description of the feature was also no longer accurate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously only possible per context, new option called imapfolder.
(closes issue #14298)
Reported by: jablko
Patches:
patch-200906202 uploaded by jablko (license 675)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This option can be used to enable #exec support in the asterisk.conf configuration file.
(closes issue #16260)
Reported by: atis
Patches:
exec_includes.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication (Collect calls)
The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was
variable and did not allow for the easy addition of more options.
The earlier 'C' prefix character for reverse charge indiation would
conflict with the a-d DTMF digits if ISDN uses them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change causes answertime to be correct even if the called channel hangs up during an announcement triggered by the A() option.
(closes issue #15936)
Reported by: falves11
Patches:
dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
dial-caller-answer1.diff uploaded by mnicholson (license 96)
Tested by: falves11, mnicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a configuration option to SIP peers, unsolicited_mailbox, which
configures a virtual mailbox to use for received new/old MWI information. This
virtual mailbox can then be used by any device supporting MWI.
(closes issue #13028)
Reported by: AsteriskRocks
Patches:
bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the second commit for this and documents the text stream using the configured
IP address and fixes a bug in the original patch where the UDPTL stream would also
use the different IP address.
(closes issue #14729)
Reported by: _brent_
Patches:
media_address.patch uploaded by brent (license 388)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ability has been added to both manager command SIPnotify as well as console
command sip notify. Message body is stored in the "Content" variable. An
example is present in sip_notify.conf.
(closes issue #13926)
Reported by: jthurman
Patches:
sip-notify-svn189463.diff uploaded by gareth (license 208)
Tested by: gareth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PRI channels can no longer change the channel name if a different B
channel is selected during call negotiation. To prevent using the channel
name to infer what B channel a call is using and to avoid name collisions,
the channel name format is changed.
The new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
