This simply pulls in the changes that were breaking from the CHANGES file
and updates a few other areas accordingly. It also removes the 10 => 11
notes, which are traditionally removed from each major version and stored
in the appropriate UPGRADE-X.txt file.
........

Merged revisions 398100 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398101 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ill conceived chan_agent is no more.  It is now replaced by
app_agent_pool.

Agents login using the AgentLogin() application as before.  The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan.  (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)

Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()

Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001

Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
   basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
   the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.

To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support.  The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback.  The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.

(closes issue ASTERISK-21554)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2657/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.

(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
    This means CDRs track well with what an actual channel is doing - which
    is useful in transfer scenarios (which were previously difficult to pin
    down). It does, however, mean that CDRs cannot be 'fooled'. Previous
    behavior in Asterisk allowed for CDR applications, channels, and other
    properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
    be what everyone wants, but it is a defined behavior and as such, it is
    predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
    changes have been made to ResetCDR and ForkCDR in particular. Many of the
    options for these two applications no longer made any sense with the new
    framework and the (slightly) more immutable nature of CDRs.

There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.

(closes issue ASTERISK-21196)

Review: https://reviewboard.asterisk.org/r/2486/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r386792, the ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the first caller
to continue receiving prompts while the agent is dialed, it has the side effect
of preventing the first caller from hearing the agent immediately upon
bridging. This may not be a problem for those who really want this option, but
for those who didn't care whether or not the first caller in queue heard their
position, it was an issue.

This patch disables the ability for the first caller in the queue to hear
prompts and adds a new option, announce-to-first-user, to queues.conf. Those
who the behavior can enable it by setting this value to True.

Note that if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed.

(closes issue ASTERISK-21782)
Reported by: Remi Quezada
........

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........

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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

Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-21151)


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

This allows Asterisk to start without having to specify the
LD_LIBRARY_PATH. This can be disabled by passing --disable-rpath to
configure.

(closes issue ASTERISK-20407)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2132/
........

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file:///srv/subversion/repos/asterisk/trunk

................
  r379021 | dlee | 2013-01-14 09:29:22 -0600 (Mon, 14 Jan 2013) | 15 lines
  
  Fix XML encoding of 'identity display' in NOTIFY messages, continued.
  
  When r378933 was merged into 1.8, it should have also escaped
  remote_display, since it will have the same XML encoding problem when
  the caller/callee roles are reversed.
  
  (closes issue ABE-2902)
  Reported by: Guenther Kelleter
  ........
  
  Merged revisions 379001 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 379020 from http://svn.asterisk.org/svn/asterisk/branches/11
................
  r379023 | dlee | 2013-01-14 09:58:01 -0600 (Mon, 14 Jan 2013) | 20 lines
  
  Masquerades are an insane implementation detail within Asterisk. It generates
  a number of useless and confusing events, and manipulates channels in a way
  that semantically doesn't make sense. I've given a fairly thorough review of
  masquerade code and its usage on the wiki at
  https://wiki.asterisk.org/wiki/x/IwBRAQ.
  
  While ultimately it makes the most sense to abandon masquerades altogether,
  it will take some time to completely irradicate. Even then, there may always
  be code that's not worth rewriting to get rid of the masquerade.
  
  This patch does two things to make masquerades slightly less insane:
   * When swapping the names of the original and clone channel, only emit a
     single rename event of original -> original<ZOMBIE>. The original code
     issued three rename events to accomplish the same end.
   * In addition to swapping the names of the channels, also swap their
     uniqueid's. This allows the 'Uniqueid' field to be used as a stable
     identifier for a channel from and external interface, such as AMI.
  
  Review: https://reviewboard.asterisk.org/r/2266/
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379032 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.

While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.

This patch does two things to make masquerades slightly less insane:
 * When swapping the names of the original and clone channel, only emit a
   single rename event of original -> original<ZOMBIE>. The original code
   issued three rename events to accomplish the same end.
 * In addition to swapping the names of the channels, also swap their
   uniqueid's. This allows the 'Uniqueid' field to be used as a stable
   identifier for a channel from and external interface, such as AMI.

Review: https://reviewboard.asterisk.org/r/2266/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

................
  r378688 | rmudgett | 2013-01-08 17:44:26 -0600 (Tue, 08 Jan 2013) | 35 lines
  
  app_queue: Fix multiple calls to a queue member that is in only one queue.
  
  When ringinuse=no queue members can receive more than one call if these
  calls happen at nearly the same time.
  
  * Fix so a queue member does not receive more than one call from a queue.
  
  NOTE: This fix does not prevent multiple calls to a member if the member
  is in more than one queue.
  
  * Did some refactoring to eliminate some code redundancy.
  
  (issue ASTERISK-16115)
  Reported by: nik600
  Patches:
        jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
        Modified
  
  * Revert the -r341580 and -r341599 changes adding the queues.conf
  check_state_unknown option as it was added in an attempt to fix this
  problem.  The fix did not need to be optional.  The fix should not have
  tried to explicitly set the device state.  Setting the device state by
  something other than the device introduces a race condition.  I also could
  not see how the change would be effective other than delaying the
  app_queue code long enough for the device state to propagate to app_queue.
  ........
  
  Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
  Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11
................
  r378691 | rmudgett | 2013-01-08 18:05:35 -0600 (Tue, 08 Jan 2013) | 10 lines
  
  app_queue: Fix incorrect assertion.
  
  (issue ASTERISK-16115)
  ........
  
  Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
  Merged revisions 378690 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.

* Fix so a queue member does not receive more than one call from a queue.

NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.

* Did some refactoring to eliminate some code redundancy.

(issue ASTERISK-16115)
Reported by: nik600
Patches:
      jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
      Modified

* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem.  The fix did not need to be optional.  The fix should not have
tried to explicitly set the device state.  Setting the device state by
something other than the device introduces a race condition.  I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
........

Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

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file:///srv/subversion/repos/asterisk/trunk

........
  r378063 | jrose | 2012-12-14 16:34:18 -0600 (Fri, 14 Dec 2012) | 8 lines
  
  Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
  
  BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
  does. In addition, the BRIDGE_FEATURES variable would not apply features to
  the proper party based on whether the feature option letter was in caps or
  in lowercase (both ways would apply it to the caller). Now uppercase applies
  to the caller while lowercase applies to the callee (like with the dial option)
........
  r378064 | rmudgett | 2012-12-14 16:45:03 -0600 (Fri, 14 Dec 2012) | 4 lines
  
  chan_agent: Remove some duplicated code.
  
  No need to check for an agent twice.  Santa does that.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378066 65c4cc65-6c06-0410-ace0-fbb531ad65f3

BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

................
  r376691 | rmudgett | 2012-11-27 18:13:10 -0600 (Tue, 27 Nov 2012) | 39 lines
  
  Fix extension matching with the '-' char.
  
  The '-' char is supposed to be ignored by the dialplan extension matching.
  Unfortunately, it's treatment is not handled consistently throughout the
  extension matching code.
  
  * Made the old exten matching code consistently ignore '-' chars.
  
  * Made the old exten matching code consistently handle case in the
  matching.
  
  * Made ignore empty character sets.
  
  * Fixed ast_extension_cmp() to return -1, 0, or 1 as documented.  The only
  user of it in pbx_lua.c was testing for -1.  It was originally returning
  the strcmp() value for less than which is not usually going to be -1.
  
  * Fix character set sorting if the sets have the same number of characters
  and start with the same character.  Character set [0-9] now sorts before
  [02-9a] as originally intended.
  
  * Updated some extension label and priority already in use warnings to
  also indicate if the extension is aliased.
  
  (closes issue ASTERISK-19205)
  Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/2201/
  ........
  
  Merged revisions 376688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 376689 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
  Merged revisions 376690 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376698 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.

* Made the old exten matching code consistently ignore '-' chars.

* Made the old exten matching code consistently handle case in the
matching.

* Made ignore empty character sets.

* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented.  The only
user of it in pbx_lua.c was testing for -1.  It was originally returning
the strcmp() value for less than which is not usually going to be -1.

* Fix character set sorting if the sets have the same number of characters
and start with the same character.  Character set [0-9] now sorts before
[02-9a] as originally intended.

* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.

(closes issue ASTERISK-19205)
Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2201/
........

Merged revisions 376688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The change in question was added to improve compliance with RFC3261, but at
the time of commit, it wasn't adequately documented in the UPGRADE notes.

(closes issue ASTERISK-20561)
Reported by: Deniz
Review: https://reviewboard.asterisk.org/r/2177/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to inconsistencies in how variable names were evaluated, the
decision was made to make all evaluations case-sensitive. See the
UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity
for more details.

(closes issue ASTERISK-20163)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2160


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a caller enters a queue and no queue member answers the call, the current
behaviour can be a little odd depending on the paused status of the queue
members.  If any queue member is paused, but not all, the CDR disposition
will be BUSY.  If all queue members are paused, then the CDR disposition is
based instead on the disposition of the call prior to entering the Queue.

This patch modifies the behaviour in the following ways:
* If no queue members are paused, the CDR disposition is whatever the
  disposition was prior to going into Queue.  If the call was answered this
  will be ANSWERED; otherwise, it is NO ANSWER.
* If some queue members are pused, the CDR result is NO ANSWER. (This is a
  change in behaviour, as the result would previously have been BUSY)
* If all queue members are paused, the CDR result is whatever the result was
  prior to going into Queue.  This is the same as the behaviour prior to this
  patch.
* If the caller hangs up, times out, or presses '*' with the 'h' option, the
  CDR disposition is again not set and is dependent on whether or not the
  caller was Answered prior to entering Queue.

This patch was based on one provided by Thomas Arimont, but has been modified
to accomodate findings by the reviewers.

Review: https://reviewboard.asterisk.org/r/2064/

(closes issue AST-906)
Reported by: Thomas Arimont

(closes issue ASTERISK-17776)
Reported by: Attila Megyeri



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds UPGRADE notes describing behavioral changes to rrmemory strategy caused by
375216

(issue AST-989)
Reported by: Thomas Arimont


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second.  Dial, ExternalIVR, and SendDTMF.

* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'.  The 'w' pauses dialing for half a
second.  The 'W' pauses dialing for one second.

* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.

(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
      jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
      Expanded patch to add support in chan_dahdi.
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue AST-969)
Reported by John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding UPGRADE.txt entry for r372148

(issue AST-946)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372149 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Matt Jordan  informed me that it was more appropriate to use an
astman_send_ack here instead of making an event response. I've also
used this opportunity to update UPGRADE.txt to mention this change
in behavior.

(issue AST-969)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370382 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.

(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.

This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.

You may ask yourself though where the name motif comes from... and I would say to you... music!

motif: a perceivable or salient recurring fragment or succession of notes

Sorta like a jingle!

Review: https://reviewboard.asterisk.org/r/1917/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ANI2 information is now parsed out of SIP From headers when present in
the oli, isup-oli, and ss7-oli parameters and is available via the
CALLERID(ani2) dialplan function.

(closes issue ASTERISK-19912)
Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19322)

(closes issue ASTERISK-19875)
Reported by: call
........

Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.

* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.

Review: https://reviewboard.asterisk.org/r/1829/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.

(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1855/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3