Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5

Currently, if any custom ring cadences are specified, they are
appended to the array of cadences from wherever we left off
last time. This works properly the first time, but on subsequent
dahdi restarts, it means that the existing cadences are left
alone and (most likely) the same cadences are then re-added
afterwards. In short order, the cadence array gets maxed out
and the user begins seeing warnings that the array is full
and no more cadences may be added.

This buggy behavior persists until Asterisk is completely
restarted; however, if and when dahdi restart is run again,
then the same problem is reintroduced.

This fixes this behavior so that cadence parsing is more
idempotent, that is so running dahdi restart multiple times
starts adding cadences from the beginning, rather than from
wherever the last cadence was added.

As before, it is still not possible to revert to the default
cadences by simply removing all cadences in this manner, nor
is it possible to delete existing cadences. However, this
does make it possible to update existing cadences, which
was not possible before, and also ensures that the cadences
remain unchanged if the config remains unchanged.

ASTERISK-29990 #close

Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d

Correct typos of the following word families:

password
excludes
undesirable
checksums
through
screening
interpreting
database
causes
initiation
member
busydetect
defined
severely
throughput
recognized
counter
require
indefinitely
accounts

ASTERISK-29714

Change-Id: Ie8f2a7b274a162dd627ee6a2165f5e8a3876527e

This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.

In particular the behaviours that this impacts include:

 - FGC-CAMA did not work at all when using MF signaling. Modified the
   switch case block to send calls to the correct part of the
   signaling-handling state machine.

 - For FGC-CAMA operation, the delay between called number ST and
   second wink for ANI spill has been made configurable; previously
   all calls were made to wait for one full second.

 - After the ANI spill, previous behavior was to require a 'ST' tone
   to advance the call.  This has been changed to allow 'STP' 'ST2P'
   or 'ST3P' as well, for compatibility with ANI-D.

 - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.

 - For calls with an ANI failure, No. 1 Crossbar switches will send
   forward a single-digit failure code, with no calling number digits
   and no ST pulse to terminate the spill.  I've made the ANI timeout
   configurable so to reduce dead air time on calls with ANI fail.

 - ANI info digits configurable.  Modern digital switches will send 2
   digits, but ANI-B sends only a single info digit.  This caused the
   ANI reported by Asterisk to be misaligned.

 - Changed a confusing log message to be more informative.

ASTERISK-29518

Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256

Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers.  It has three timeout constants that this commit converts to
channel-level configuration options:

* firstdigit_timeout: Default time (ms) to detect first digit

* interdigit_timeout: Default time (ms) to detect following digits

* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match.  This happens when the dialed digits match a number in the current
context but are also the prefix of another number.

Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Support for referring to DAHDI channels by logical names was added in
(FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support
of refering to channels by name.

While technically usable, it has never been properly supported in
dahdi-tools, as using it would require many changes at the Asterisk
level. Instead logical mapping was added at the kernel level.

Thus it seems that refering to DAHDI channels by name is not really used
by anyone, and therefore should probably be removed.

Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485

Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6

The new option allows the channel driver's faxdetect option to timeout on
a call after the specified number of seconds into a call.  The new feature
is disabled if the timeout is set to zero.  The option is disabled by
default.

* Don't clear dsp_features after passing them to the dsp code in
my_pri_ss7_open_media().  We should still remember them especially for the
new faxdetect_timeout option.

ASTERISK-26214
Reported by: Richard Mudgett

Change-Id: Ieffd3fe788788d56282844774365546dce8ac810

ASTERISK-25034
Reported by: Richard Mudgett

Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30

Some telco switches occasionally ignore ISDN RESTART requests.  The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request.  If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco.  The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.

chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code.  Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards.  I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.

* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.

ASTERISK-25034 #close
Reported by: Richard Mudgett

Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.

Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP.  It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed.  This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.

NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.

The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.

This commit is a merge of the two patches indicated below.

ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
      pri-4.diff (license #6302) patch uploaded by Pavel Troller
      jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett

Review: https://reviewboard.asterisk.org/r/3633/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* SS7 support now requires libss7 v2.0 or later.  The new libss7 is not
backwards compatible.

* Added SS7 support for connected line and redirecting.

* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.

* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.

* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.

Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.

SS7-27 #close
Reported by: adomjan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If ignore_failed_channels is set to "true" for a channel, the channel
will continue to be configured even if configuring it has failed.

This allows Asterisk to start before all the DAHDI initialization is
done and thus not force the starting order dahdi -> asterisk.

Review: https://reviewboard.asterisk.org/r/3063/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change is in preparation for external MWI support.

Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context.  The
only exception is the legacy hasvoicemail users.conf option.  The legacy
option will only work for app_voicemail mailboxes.  The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.

chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier.  chan_sip just stored and
compared the two components.  chan_dahdi actually used the box
information.

The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number.  As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.

Review: https://reviewboard.asterisk.org/r/3072/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with
documentation in chan_dahdi.conf.sample.

(closes issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
      jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Changing text in chan_dahdi.conf sample to be accurate.

(issue ASTERISK-22308)
(closes issue ASTERISK-22308)
Reported By: Malcolm Davenport
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* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific.  If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.

* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge.  The BRIDGEPEER value can have a maximum of 10
peers listed.  Any more peers in the bridge will not be included in the
list.  BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.

* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.

* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed.  The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.

* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature.  Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.

(closes issue ASTERISK-21555)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2582/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-21151)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.

Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio.  However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.

ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.

(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This contains an edited version of the patch originally created by John
Bigelow.

(closes issue ASTERISK-14435)
Reported by: John Bigelow
Patches:
      buffers.patch (license #5091) patch uploaded by John Bigelow
      0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell
      Modified
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374730 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds named calledgroups/pickupgroups to Asterisk.  Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation.  However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.

Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup".  This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup".  Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.

Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.

Review: https://reviewboard.asterisk.org/r/2043

Uploaded by:
	Guenther Kelleter(license #6372)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Several telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls.  Incoming
calls could fail as well because the alarm processing is handled by a
different code path than the Q.931 messages.

* Add the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down.  This option can be configured by span
while the similar DAHDI driver teignorered=1 option is system wide.  This
option unlike layer2_persistence does not require libpri v1.4.13 or newer.

Related to JIRA AST-598

JIRA ABE-2845
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added new chan_dahdi.conf colp_send option parameter to block connected
line updates per span.

(closes issue ASTERISK-17025)
Reported by: Michael Smith


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered.  If dialtone is
detected, the call is hung up.
no:       Disabled. (Default)
yes:      Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always:   Look for dialtone for the entire call.  Dialtone may return
          if the far end hangs up first.

dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always

(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper

Review: https://reviewboard.asterisk.org/r/1737/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add warning about the SIP allowguest option in context public.

(closes issue ASTERISK-14122)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/719/
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines
  
  Merged revisions 332264 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines
    
    Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
    
    France Telecom brings layer 2 and layer 1 down on BRI lines when the line
    is idle.  When layer 1 goes down Asterisk cannot make outgoing calls and
    the HA8 and HB8 cards also get IRQ misses.
    
    The inability to make outgoing calls is because the line is in red alarm
    and Asterisk will not make calls over a line it considers unavailable.
    The IRQ misses for the HA8 and HB8 card are because the hardware is
    switching clock sources from the line which just brought layer 1 down to
    internal timing.
    
    There is a DAHDI option for the B410P card to not tell Asterisk that layer
    1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
    teignored=1".  There is a similar DAHDI option for the HA8 and HB8 cards:
    "modprobe wctdm24xxp bri_teignored=1".  Unfortunately that will not clear
    up the IRQ misses when the telco brings layer 1 down.
    
    * Add layer 2 persistence option to customize the layer 2 behavior on BRI
    PTMP lines.  The new option has three settings: 1) Use libpri default
    layer 2 setting.  2) Keep layer 2 up.  Bring layer 2 back up when the peer
    brings it down.  3) Leave layer 2 down when the peer brings it down.
    Layer 2 will be brought up as needed for outgoing calls.
    
    JIRA AST-598
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  r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines
  
  Merged revisions 329203 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines
    
    Document parkinglot in chan_dahdi.conf.sample.
    
    * Document existing feature in chan_dahdi.conf.sample.
    
    * Remove some dead code related to the parkinglot option.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding links to http(s)://wiki.asterisk.org



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

..........
  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
..........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r295869 | seanbright | 2010-11-22 15:03:49 -0500 (Mon, 22 Nov 2010) | 9 lines
  
  Merged revisions 295868 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov 2010) | 2 lines
    
    Change some documentation to suggest dahdi_monitor instead of ztmonitor.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) | 1 line
  
  Update chan_dahdi.conf.sample to reflect new libpri T309 default value.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286427 65c4cc65-6c06-0410-ace0-fbb531ad65f3