git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch, originally submitted by jozza, enables custom modules to send actions to AMI
and receive messages from AMI via a hook interface. Included is a simple test module to
illustrate the interface.

(closes issue #14635)
Reported by: jozza

Review: https://reviewboard.asterisk.org/r/412/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a sequence field to CDRs that can be combined with the linkedid or uniqueid field to uniquely identify a CDR.

(closes issue #15180)
Reported by: Nick_Lewis
Patches:
      cdr-sequence10.diff uploaded by mnicholson (license 96)
Tested by: mnicholson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add support for using a hint when configuring a state interface using the format hint:<extension>@<context>.

(closes issue #15168)
Reported by: p_lindheimer
Patches:
      queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14517)
(SWP-109)
Reported by: asgaroth
Patches:
      bug_14517.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy, dougm, qwell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227372 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app_controlplayback outputs a warning, when in fact it is normal.

(closes issue #16071)
Reported by: atis
Patches:
      controlplayback_warning.patch uploaded by atis (license 242)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update the extensions.conf.sample [stdexten] context so that we use the 
variable instead of requiring it to be passed explicitly. Also updated uses of
the [stdexten] context throughout.

(closes issue #15858)
Reported by: pprindeville
Patches:
      stdexten-context-update.txt uploaded by lmadsen (license 10)
Tested by: pprindeville

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227361 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines
  
  Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls.
  
  This is the relevant portion of asterisk/trunk -r226648
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227277 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16120)
Reported by: jsmith


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The functions needed doesn't exist in Speex 1.05 which is what a lot of distros use.
1.2 seems to have been in beta status for years, and does include the sexy functions needed for func_speex to work.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines
  
  Fix a bug where an RPID header could be generated with a blank username in the URI.
  
  (closes issue #15909)
  Reported by: kobaz
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15857)
Reported by: pprindeville
Patches:
      stdexten.patch#2 uploaded by pprindeville (license 347)
Tested by: pprindeville

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227162 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines

Use proper response code when violating Contact ACL's.

https://reviewboard.asterisk.org/r/415/

Thanks kpfleming for a quick review.
(EDVX-003)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12950)
 Reported by: alea-soluciones
 Patches: 
       ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514)
 Tested by: alea-soluciones, adomjan, urtho, nahuelgreco


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.

(closes issue #15152)
Reported by: palbrecht

Review: https://reviewboard.asterisk.org/r/420/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.

(closes issue #15152)
Reported by: palbrecht

Review: https://reviewboard.asterisk.org/r/420/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226973 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226970 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
  
  Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
  while the called party had not yet answered.
  
  This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
  file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
  
  (closes issue #14674)
  Reported by: ulogic
  Patches:
        bug14674.patch uploaded by jpeeler (license 325)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226890 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work.  This has
not been an issue until the advent of PTMP NT mode.  Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.

As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>

Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212

This will work with the following restrictions:
*  The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
*  Each device/phone can only have one number.  No shared MSN's.
*  The phones/devices probably should not use subaddressing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines
  
  Don't allow two separate instances of safe_asterisk when restarting from the init script.
  (closes issue #14562)
   Reported by: davidw
   Patches: 
         Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
         Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780)
   Tested by: davidw
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r226736 | dvossel | 2009-11-02 09:31:02 -0600 (Mon, 02 Nov 2009) | 5 lines
  
  fixes crash on iterator_destroy on uninitialized iterator
  
  (closes issue #16162)
  Reported by: krn
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r226688 | dvossel | 2009-11-02 09:16:30 -0600 (Mon, 02 Nov 2009) | 5 lines
  
  changes calltoken debug messages from LOG_NOTICE to LOG_DEBUG like they are supposed to be
  
  (closes issue #16144)
  Reported by: aragon
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13385)
Reported by: adomjan
Patches:
      sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487)
      CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487)
      chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487)
      sip-q850-hangupcause1.diff uploaded by mnicholson (license 96)
Tested by: adomjan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3

*  Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split)
*  Make sure the outgoing flag is cleared if a new channel fails to get
created for outgoing calls.
*  Remove some unused flags since sig_pri was split.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is a side project I've been poking at this week.  The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together.  There is a ton of stuff to write about, so this will
just continue to evolve over time.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines
  
  Add an option to enabling passing music on hold start and stop requests through instead of
  acting on them in chan_local.
  
  (closes issue #14709)
  Reported by: dimas
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
Just remove that. No pattern at all always matches.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226453 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
  
  Update documentation in sip.conf.sample.
  
  Update the documentation in sip.conf.sample in order to make it more clear
  that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
  is only used to stop Asterisk from generating a reINVITE, but does not stop
  it from accepting them if necessary.
  
  (closes issue #15644)
  Reported by: lmadsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines
  
  Update CALLINGSUBADDR channel variable documentation.
  
  (closes issue #15734)
  Reported by: alecdavis
  Patches:
        channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines
  
  Fix documentation (pointed out by TheDavidFactor on #-dev)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226305 65c4cc65-6c06-0410-ace0-fbb531ad65f3

/var/run would be cleaned on startup on most systems anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines
  
  Manager output is not always NULL-terminated, so force a NULL at the end of the filestream.
  (closes issue #15495)
   Reported by: pdf
   Patches: 
         20090916__issue15495.diff.txt uploaded by tilghman (license 14)
   Tested by: pdf
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226159 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226099 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