asterisk/asterisk.git
10 years agoFix some build issues on Solaris.
Jason Parker [Tue, 3 Nov 2009 19:59:46 +0000 (19:59 +0000)]
Fix some build issues on Solaris.

(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

10 years agoChange warning message to debug message.
Leif Madsen [Tue, 3 Nov 2009 19:48:53 +0000 (19:48 +0000)]
Change warning message to debug message.

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

10 years agoAdditional fixes to the extensions.conf.sample file.
Leif Madsen [Tue, 3 Nov 2009 19:25:18 +0000 (19:25 +0000)]
Additional fixes to the extensions.conf.sample file.

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

10 years agoFixed a spelling error in the q850 reason header option in the output of sip show...
Matthew Nicholson [Tue, 3 Nov 2009 18:22:28 +0000 (18:22 +0000)]
Fixed a spelling error in the q850 reason header option in the output of sip show settings.

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

10 years agoRecorded merge of revisions 227275 via svnmerge from
Richard Mudgett [Tue, 3 Nov 2009 17:58:38 +0000 (17:58 +0000)]
Recorded merge of revisions 227275 via svnmerge from
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

10 years agoCode guidelines fixes only
Tilghman Lesher [Tue, 3 Nov 2009 17:56:41 +0000 (17:56 +0000)]
Code guidelines fixes only

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

10 years agouser.conf entries in SIP were not having their peer type set.
David Vossel [Tue, 3 Nov 2009 17:12:52 +0000 (17:12 +0000)]
user.conf entries in SIP were not having their peer type set.

(closes issue #16120)
Reported by: jsmith

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

10 years agoAdding some clarifications to func_speex doxygen docs.
Olle Johansson [Tue, 3 Nov 2009 16:56:48 +0000 (16:56 +0000)]
Adding some clarifications to func_speex doxygen docs.

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

10 years agoMerged revisions 227166 via svnmerge from
Joshua Colp [Tue, 3 Nov 2009 15:37:08 +0000 (15:37 +0000)]
Merged revisions 227166 via svnmerge from
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

10 years agoUpdate extensions.conf.sample file to fix incorrect extensions.
Leif Madsen [Tue, 3 Nov 2009 15:19:47 +0000 (15:19 +0000)]
Update extensions.conf.sample file to fix incorrect extensions.

(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

10 years agoMerged revisions 227088 via svnmerge from
Olle Johansson [Tue, 3 Nov 2009 11:11:15 +0000 (11:11 +0000)]
Merged revisions 227088 via svnmerge from
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

10 years agoAdd PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
Tilghman Lesher [Mon, 2 Nov 2009 22:29:19 +0000 (22:29 +0000)]
Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
(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

10 years agoSIP channel name uniqueness
David Brooks [Mon, 2 Nov 2009 20:59:37 +0000 (20:59 +0000)]
SIP channel name uniqueness

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

10 years agoSIP channel name uniqueness
David Brooks [Mon, 2 Nov 2009 20:57:45 +0000 (20:57 +0000)]
SIP channel name uniqueness

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

10 years agoAdding external reference for doxygen
Olle Johansson [Mon, 2 Nov 2009 20:43:52 +0000 (20:43 +0000)]
Adding external reference for doxygen

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

10 years agoMerged revisions 226889 via svnmerge from
Joshua Colp [Mon, 2 Nov 2009 18:08:54 +0000 (18:08 +0000)]
Merged revisions 226889 via svnmerge from
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

10 years agoDAHDI ISDN channel names will not allow device state to work. (Interim solution.)
Richard Mudgett [Mon, 2 Nov 2009 17:34:22 +0000 (17:34 +0000)]
DAHDI ISDN channel names will not allow device state to work.  (Interim solution.)

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

10 years agoMerged revisions 226811 via svnmerge from
Tilghman Lesher [Mon, 2 Nov 2009 17:15:31 +0000 (17:15 +0000)]
Merged revisions 226811 via svnmerge from
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

10 years agoBlocked revisions 226736 via svnmerge
David Vossel [Mon, 2 Nov 2009 15:34:37 +0000 (15:34 +0000)]
Blocked revisions 226736 via svnmerge

........
  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

10 years agoBlocked revisions 226688 via svnmerge
David Vossel [Mon, 2 Nov 2009 15:17:04 +0000 (15:17 +0000)]
Blocked revisions 226688 via svnmerge

........
  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

10 years agoThis patch adds support for a draft proposal for adding Q.850 reason headers to sip...
Matthew Nicholson [Mon, 2 Nov 2009 14:57:11 +0000 (14:57 +0000)]
This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages.

(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

10 years agoCleanup some flags on DAHDI PRI channel hangup.
Richard Mudgett [Fri, 30 Oct 2009 23:26:41 +0000 (23:26 +0000)]
Cleanup some flags on DAHDI PRI channel hangup.

*  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

10 years agoAdd an "Asterisk Architecture Overview" section to the doxygen documentation.
Russell Bryant [Fri, 30 Oct 2009 04:08:39 +0000 (04:08 +0000)]
Add an "Asterisk Architecture Overview" section to the doxygen documentation.

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

10 years agoMerged revisions 226531 via svnmerge from
Joshua Colp [Thu, 29 Oct 2009 18:13:42 +0000 (18:13 +0000)]
Merged revisions 226531 via svnmerge from
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

10 years agoDoxygen documentation update
Olle Johansson [Thu, 29 Oct 2009 12:20:16 +0000 (12:20 +0000)]
Doxygen documentation update

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

10 years agoremove empty awk pattern (//)
Tzafrir Cohen [Wed, 28 Oct 2009 20:50:52 +0000 (20:50 +0000)]
remove empty awk pattern (//)

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

10 years agoMerged revisions 226382 via svnmerge from
Leif Madsen [Wed, 28 Oct 2009 20:11:07 +0000 (20:11 +0000)]
Merged revisions 226382 via svnmerge from
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

10 years agoMerged revisions 226377 via svnmerge from
Leif Madsen [Wed, 28 Oct 2009 19:50:00 +0000 (19:50 +0000)]
Merged revisions 226377 via svnmerge from
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

10 years agoMerged revisions 226304 via svnmerge from
Tilghman Lesher [Wed, 28 Oct 2009 18:04:05 +0000 (18:04 +0000)]
Merged revisions 226304 via svnmerge from
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

10 years agoRemove extra cleanup in case we have more than one Asterisk.
Tzafrir Cohen [Wed, 28 Oct 2009 08:47:59 +0000 (08:47 +0000)]
Remove extra cleanup in case we have more than one Asterisk.

/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

10 years agoanother variation of the upstart script
Tzafrir Cohen [Tue, 27 Oct 2009 22:10:38 +0000 (22:10 +0000)]
another variation of the upstart script

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

10 years agoAdding compile time flags for Snow Leopard, Leopard and some other animals
Olle Johansson [Tue, 27 Oct 2009 21:03:22 +0000 (21:03 +0000)]
Adding compile time flags for Snow Leopard, Leopard and some other animals

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

10 years agoMerged revisions 226138 via svnmerge from
Tilghman Lesher [Tue, 27 Oct 2009 20:22:07 +0000 (20:22 +0000)]
Merged revisions 226138 via svnmerge from
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

10 years agoDon't prepend the URI prefix to the post directory
Terry Wilson [Tue, 27 Oct 2009 16:48:54 +0000 (16:48 +0000)]
Don't prepend the URI prefix to the post directory

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226099 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd support for receiving unsolicited MWI NOTIFY messages.
Joshua Colp [Tue, 27 Oct 2009 13:30:27 +0000 (13:30 +0000)]
Add support for receiving unsolicited MWI NOTIFY messages.

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

10 years agodetect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi
Tzafrir Cohen [Mon, 26 Oct 2009 22:46:09 +0000 (22:46 +0000)]
detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi

* Set OSARCH to linux-gnu even if host_os is linux-gnueabi
* When checking if we are Linux, check OSARCH rather than host_os

The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than
'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case.

OSARCH is tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH rather
than $host_os .

See also: http://wiki.debian.org/ArmEabiPort

Merged revisions 225957 via svnmerge from
http://svn.digium.com/svn/asterisk/branches/1.4

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix building in REF_DEBUG mode.
Kevin P. Fleming [Mon, 26 Oct 2009 22:04:04 +0000 (22:04 +0000)]
Fix building in REF_DEBUG mode.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoCorrect broken logic from revision 225405.
Kevin P. Fleming [Mon, 26 Oct 2009 22:03:29 +0000 (22:03 +0000)]
Correct broken logic from revision 225405.

The code committed in revision 225405 was broken; instead of removing the unreference code,
the logic used to decide when to do it should have been reversed. This patch corrects the
situation, and makes reference counting work properly again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225955 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoACL check not present for verifying SIP INVITEs
Jeff Peeler [Mon, 26 Oct 2009 19:40:26 +0000 (19:40 +0000)]
ACL check not present for verifying SIP INVITEs

The ACL check in check_peer_ok was missing and has now been restored. The
missing check allowed for calls to be made on prohibited networks where an ACL
was defined in sip.conf and the allowguest option was set to off. See the AST
security advisory below for more information.

Merge code associated with AST-2009-007.

(closes issue #16091)
Reported by: thom4fun

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake conditionals create previous code when libpri/ss7 are present.
Richard Mudgett [Mon, 26 Oct 2009 16:07:09 +0000 (16:07 +0000)]
Make conditionals create previous code when libpri/ss7 are present.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225872 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agospan numbers in pri debug / error messages
Tzafrir Cohen [Mon, 26 Oct 2009 13:29:54 +0000 (13:29 +0000)]
span numbers in pri debug / error messages

Prefix PRI trace messages with the span number. This makes the trace
readable even when you have a multi-port device.

(closes issue #15054)
Reported by: tzafrir
Patches:
      dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRe-arange code a bit to build in dev-mode without ss7
Tzafrir Cohen [Mon, 26 Oct 2009 11:34:06 +0000 (11:34 +0000)]
Re-arange code a bit to build in dev-mode without ss7

No change of functionality here. Just localized a variable and indented
code into blocks.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225803 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake chan_dahdi build even without PRI / SS7
Tzafrir Cohen [Mon, 26 Oct 2009 09:40:49 +0000 (09:40 +0000)]
Make chan_dahdi build even without PRI / SS7

(Note: still some strange build warnings without SS7 in dev-mode)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoImprove performance of pedantic mode dialog searching in chan_sip.
Kevin P. Fleming [Sat, 24 Oct 2009 14:40:37 +0000 (14:40 +0000)]
Improve performance of pedantic mode dialog searching in chan_sip.

This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support
to make pedantic mode dialog searching in find_call() not require a linear search
of all dialogs in the list of dialogs. This patch does *not* change the dialog
matching logic (more on that later), just improves the searching performance.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.
Richard Mudgett [Fri, 23 Oct 2009 16:57:33 +0000 (16:57 +0000)]
Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.

* 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

10 years agoOptionally build and install the sample AGIs in the agi/ directory.
Sean Bright [Fri, 23 Oct 2009 16:40:30 +0000 (16:40 +0000)]
Optionally build and install the sample AGIs in the agi/ directory.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225690 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFixes an iterator memory leak and uninitialized memory
David Vossel [Fri, 23 Oct 2009 14:41:50 +0000 (14:41 +0000)]
Fixes an iterator memory leak and uninitialized memory

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225650 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225581 via svnmerge from
Kevin P. Fleming [Fri, 23 Oct 2009 14:02:42 +0000 (14:02 +0000)]
Merged revisions 225581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines

  Don't force menuselect.makeopts to be rebuilt on every build.

  For some reason the menuselect.makeopts file was listed as PHONY in the Makefile,
  resulting in 'make' needing to rebuild it for every build. This then resulted in
  the embedded module rules being rebuilt on every build, which can be slow and is
  unnecessary.

  This patch fixes the problem by properly allowing 'make' to know when the
  menuselect.makeopts file needs to be rebuilt (defining the proper dependencies).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoUpdate README documentation.
Leif Madsen [Thu, 22 Oct 2009 22:24:03 +0000 (22:24 +0000)]
Update README documentation.
Update the README documentation to correctly describe which CLI command you should
use when attempting to get help from the CLI.

(closes issue #16064)
Reported by: thedavidfactor
Patches:
      readme.patch uploaded by thedavidfactor (license 903)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225484 via svnmerge from
Leif Madsen [Thu, 22 Oct 2009 21:52:30 +0000 (21:52 +0000)]
Merged revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines

  Clean valgrind output by suppressing false errors.
  Update valgrind.txt documentation and add valgrind.supp file in order to
  allow those who are creating valgrind output to have less false errors in
  the logfile.

  (closes issue #16007)
  Reported by: atis
  Patches:
        valgrind.txt.diff uploaded by atis (license 242)
        asterisk2.supp uploaded by atis (license 242)
  Tested by: atis, amorsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225485 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd Asterisk Git HowTo documentation.
Leif Madsen [Thu, 22 Oct 2009 21:28:44 +0000 (21:28 +0000)]
Add Asterisk Git HowTo documentation.
Added documentation on how to create a local git repository from
SVN. This documentation was added via doxygen.

(closes issue #15814)
Reported by: tzafrir
Patches:
      git-asterisk-howto uploaded by tzafrir (license 46)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoSearch for the subaddress only within the extension section of the dial string.
Richard Mudgett [Thu, 22 Oct 2009 20:07:55 +0000 (20:07 +0000)]
Search for the subaddress only within the extension section of the dial string.

Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225446 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoSIP TCP/TLS: move client connection setup/write into tcp helper thread, various relat...
David Vossel [Thu, 22 Oct 2009 19:55:51 +0000 (19:55 +0000)]
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.

        What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
  Connection setup takes awhile and before this it was being
  done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread:  Through the
  use of a packet queue and an alert pipe, the TCP helper thread
  can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
  up the tcptls_session lock.  This lock has been completely removed
  from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak:  When creating a tcptls_client the tls_cfg was alloced
  but never freed unless the tcptls_session failed to start.  Now the
  session_args for a sip client are an ao2 object which frees the
  tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
  of a client's ast_tcptls_session_args was done on the stack and
  stored as a pointer in the newly created tcptls_session.  Depending
  on the events that followed, there was a slight possibility that
  pointer could have been accessed after the stack returned.  Given
  the new changes, it is always accessed after the stack returns
  which is why I found it.

Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
  functions.  One for creating and allocating the new tcptls_session,
  and a separate one for starting and handling the new connection.
  This allowed me to create the tcptls_session, launch the helper
  thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
  This is done by using an alert pipe to wake up the thread if new
  data needs to be sent.  The thread's sip_threadinfo object contains
  the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
  accessed outside of the helper thread for every write (queuing of a
  packet).  For easy lookup, I moved the threadinfo objects from a
  linked list to an ao2_container.

(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys

(closes issue #15894)
Reported by: dvossel
Tested by: dvossel

Review: https://reviewboard.asterisk.org/r/380/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd the programs in utils/ to menuselect.
Sean Bright [Thu, 22 Oct 2009 19:33:32 +0000 (19:33 +0000)]
Add the programs in utils/ to menuselect.

Nothing in utils/ is now built by default except for astcanary.

Review: https://reviewboard.asterisk.org/r/353/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225440 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoPermit storage of voicemail secrets in a separate file, located within the spool...
Tilghman Lesher [Thu, 22 Oct 2009 19:10:04 +0000 (19:10 +0000)]
Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276)
 Reported by: klaus3000
 Patches:
       app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65)
 Tested by: jamesgolovich

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix a refcount error introduced by yesterday's OBJ_MULTIPLE commit.
Kevin P. Fleming [Thu, 22 Oct 2009 18:41:47 +0000 (18:41 +0000)]
Fix a refcount error introduced by yesterday's OBJ_MULTIPLE commit.

When an object is being unlinked from its container *and* being returned to
the caller, we do not want to decrement the reference count after unlinking
it from the container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining reference to
the object, that could result in the object being destroyed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225405 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225105 via svnmerge from
Tilghman Lesher [Thu, 22 Oct 2009 17:11:23 +0000 (17:11 +0000)]
Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines

  Fix documentation for ast_softhangup() and correct the misuse thereof.
  (closes issue #16103)
   Reported by: majorbloodnok
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd support for calling and called subaddress. Partial support for COLP subaddress.
Richard Mudgett [Thu, 22 Oct 2009 16:33:22 +0000 (16:33 +0000)]
Add support for calling and called subaddress.  Partial support for COLP subaddress.

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

10 years agoMerged revisions 225243 via svnmerge from
David Vossel [Wed, 21 Oct 2009 21:58:46 +0000 (21:58 +0000)]
Merged revisions 225243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines

  IAX2: VNAK loop caused by signaling frames with no destination call number

  It is possible for the PBX thread to queue up signaling frames before
  a destination call number is received.  This can result in signaling
  frames being sent out with no destination call number. Since recent
  versions of Asterisk require accurate destination callnumbers for all
  Full Frames, this can cause a VNAK loop to occur.  To resolve this
  no signaling frames are sent until a destination callnumber is received,
  and destination call numbers are now only required for iax_pvt matching
  when the frame is an ACK.

  Review: https://reviewboard.asterisk.org/r/413/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd 'mohsuggest' configuration option to 'sip show peer' CLI command and
Kevin P. Fleming [Wed, 21 Oct 2009 21:15:40 +0000 (21:15 +0000)]
Add 'mohsuggest' configuration option to 'sip show peer' CLI command and
SIPShowPeer AMI action.

(closes issue #15990)
Reported by: _brent_
Patches:
      sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388)

Review: https://reviewboard.asterisk.org/r/381/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFinish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to...
Kevin P. Fleming [Wed, 21 Oct 2009 21:08:47 +0000 (21:08 +0000)]
Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.

This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.

During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.

Review: https://reviewboard.asterisk.org/r/379/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225171 via svnmerge from
Russell Bryant [Wed, 21 Oct 2009 16:46:22 +0000 (16:46 +0000)]
Merged revisions 225171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines

  Revert 225169, as this doesn't account for the possibility of a list of frames.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225172 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225169 via svnmerge from
Russell Bryant [Wed, 21 Oct 2009 16:42:13 +0000 (16:42 +0000)]
Merged revisions 225169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines

  Isolate the frame returned from ast_translate().
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 225103 via svnmerge
Tilghman Lesher [Wed, 21 Oct 2009 15:46:42 +0000 (15:46 +0000)]
Blocked revisions 225103 via svnmerge

........
  r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines

  Suffix is not needed for a match
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225104 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoApparently, I don't need to specify the ".so" suffix to get a match
Tilghman Lesher [Wed, 21 Oct 2009 15:42:47 +0000 (15:42 +0000)]
Apparently, I don't need to specify the ".so" suffix to get a match

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd support for specifying the IP address to use for media streams in sip.conf
Joshua Colp [Wed, 21 Oct 2009 15:35:09 +0000 (15:35 +0000)]
Add support for specifying the IP address to use for media streams in sip.conf

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

10 years agoTurn on DENOISE filter for all conference participants.
Tilghman Lesher [Wed, 21 Oct 2009 15:21:30 +0000 (15:21 +0000)]
Turn on DENOISE filter for all conference participants.
(Fixes SWP-238)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRevert media_address commit, I'm going to roll a fix to the SDP generation in the...
Joshua Colp [Wed, 21 Oct 2009 15:04:33 +0000 (15:04 +0000)]
Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 225032 via svnmerge from
David Vossel [Wed, 21 Oct 2009 14:39:10 +0000 (14:39 +0000)]
Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines

  IAX/SIP shrinkcallerid option

  The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
  and '-' from the string.  This means values such as 555.5555 and
  test-test result in 555555 and testtest.  There are instances,
  such as Skype integration, where a specific value is passed via
  caller id that must be preserved unmodified.  This patch makes
  the shrinking of caller id optional in chan_sip and chan_iax in
  order to support such cases.  By default this option is on to
  preserve previous expected behavior.

  (closes issue #15940)
  Reported by: dimas
  Patches:
        v2-15940.patch uploaded by dimas (license 88)
        15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
  Tested by: dvossel

  Review: https://reviewboard.asterisk.org/r/408/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd support for specifying the IP address to use for media streams in sip.conf
Joshua Colp [Wed, 21 Oct 2009 13:34:49 +0000 (13:34 +0000)]
Add support for specifying the IP address to use for media streams in sip.conf

(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@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224931 via svnmerge from
Russell Bryant [Wed, 21 Oct 2009 03:09:04 +0000 (03:09 +0000)]
Merged revisions 224931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines

  Isolate frames returned from a DSP instance or codec translator.

  The reasoning for these changes are the same as what I wrote in the commit
  message for rev 222878.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake PRI_SUBCMD_xxx handling subaddress friendly.
Richard Mudgett [Wed, 21 Oct 2009 02:43:36 +0000 (02:43 +0000)]
Make PRI_SUBCMD_xxx handling subaddress friendly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224855 via svnmerge from
Tilghman Lesher [Tue, 20 Oct 2009 22:09:07 +0000 (22:09 +0000)]
Merged revisions 224855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines

  Pay attention to the return value of the manipulate function.
  While this looks like an optimization, it prevents a crash from occurring
  when used with certain audiohook callbacks (diagnosed with SVN trunk,
  backported to 1.4 to keep the source consistent across versions).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224773 via svnmerge from
Joshua Colp [Tue, 20 Oct 2009 17:47:34 +0000 (17:47 +0000)]
Merged revisions 224773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines

  Add support for relaying early media in the features attended transfer option.

  (closes issue #14828)
  Reported by: licedey
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224774 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdded information to CHANGES about the dynamic range compression feature added to...
Matthew Nicholson [Tue, 20 Oct 2009 12:44:09 +0000 (12:44 +0000)]
Added information to CHANGES about the dynamic range compression feature added to dahdi.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224738 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224670 via svnmerge from
Kevin P. Fleming [Mon, 19 Oct 2009 23:47:39 +0000 (23:47 +0000)]
Merged revisions 224670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines

  Correct timestamp calculations when RTP sample rates over 8kHz are used.

  While testing some endpoints that support 16kHz and 32kHz sample rates, some
  log messages were generated due to calc_rxstamp() computing timestamps in a way
  that produced odd results, so this patch sanitizes the result of the
  computations.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd dynamic range compression support for analog channels.
Matthew Nicholson [Mon, 19 Oct 2009 22:02:41 +0000 (22:02 +0000)]
Add dynamic range compression support for analog channels.

(closes issue AST-29)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224565 via svnmerge from
Joshua Colp [Mon, 19 Oct 2009 19:49:09 +0000 (19:49 +0000)]
Merged revisions 224565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines

  Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.

  (closes issue #14763)
  Reported by: cupotka
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove useless debugging message.
Kevin P. Fleming [Mon, 19 Oct 2009 19:40:26 +0000 (19:40 +0000)]
Remove useless debugging message.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224562 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove a completed project and add another
Tilghman Lesher [Mon, 19 Oct 2009 15:50:31 +0000 (15:50 +0000)]
Remove a completed project and add another

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd a callback to sig_pri which is called when sig_pri is going to queue a control...
Joshua Colp [Mon, 19 Oct 2009 14:32:08 +0000 (14:32 +0000)]
Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAllow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's.
Tilghman Lesher [Mon, 19 Oct 2009 00:05:56 +0000 (00:05 +0000)]
Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's.
This corrects an issue reported on the -users list.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoClarify that "forcecommit" is NOT an alias for "autocommit", but instead controls...
Tilghman Lesher [Sun, 18 Oct 2009 23:41:30 +0000 (23:41 +0000)]
Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224446 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove unnecessary typedef
Tilghman Lesher [Sat, 17 Oct 2009 16:39:37 +0000 (16:39 +0000)]
Remove unnecessary typedef

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agofix typo, sorry
Jeff Peeler [Sat, 17 Oct 2009 02:01:36 +0000 (02:01 +0000)]
fix typo, sorry

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224330 via svnmerge from
Jeff Peeler [Sat, 17 Oct 2009 01:36:08 +0000 (01:36 +0000)]
Merged revisions 224330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines

  Fix stale caller id data from being reported in AMI NewChannel event

  The problem here is that chan_dahdi is designed in such a way to set
  certain values in the dahdi_pvt only once. One of those such values
  is the configured caller id data in chan_dahdi.conf. For PRI, the
  configured caller id data could be overwritten during a call. Instead
  of saving the data and restoring, it was decided that for all non-analog
  channels it was simply best to not set the configured caller id in the
  first place and also clear it at the end of the call.

  (closes issue #15883)
  Reported by: jsmith
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 224260 via svnmerge from
Richard Mudgett [Fri, 16 Oct 2009 20:40:57 +0000 (20:40 +0000)]
Merged revisions 224260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines

  Never released PRI channels when using Busy() or Congestion() dialplan apps.

  When the Busy() or Congestion() application is used towards ISDN (an ISDN
  progress is sent), the responding ISDN Disconnect or Release may contain
  the ISDN cause user busy or one of the congestion causes.  In chan_dahdi.c
  these causes will only set the needbusy or needcongestion flags and not
  activate the softhangup procedure.  Unfortunately only the latter can
  interrupt the endless wait loop of Busy()/Congestion().

  Result: PRI channels staying in state busy for the rest of asterisk life
  or until the other end times out and forces the call to clear.

  (issue #14292)
  Reported by: tomaso
  Patches:
        disc_rel_userbusy.patch uploaded by tomaso (license 564)
        (This patch is unrelated to the issue.)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoCreate an API for adding an optional time unit onto the ends of time periods.
Tilghman Lesher [Thu, 15 Oct 2009 22:33:30 +0000 (22:33 +0000)]
Create an API for adding an optional time unit onto the ends of time periods.
Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoReadd removed ability to allow listening to one side of the call in app_chanspy
Jeff Peeler [Thu, 15 Oct 2009 15:57:14 +0000 (15:57 +0000)]
Readd removed ability to allow listening to one side of the call in app_chanspy

(Option o)

(closes issue #15675)
Reported by: john8675309
Patches:
      issue15675patchtrunk.txt uploaded by dbrooks (license 790)
Tested by: jgutierrez on users list:
 http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agochan_dahdi.conf.sample changes for DTMF CID detect
Doug Bailey [Thu, 15 Oct 2009 14:37:20 +0000 (14:37 +0000)]
chan_dahdi.conf.sample changes for DTMF CID detect

Explains new options for detecting DTMF CID on fxo lines

(issue #9096)
Reported by: fleed
Patches:
      chan_dahid_sample_config.patch uploaded by sum (license 766)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoProperly handle PUT requests for CALENDAR_WRITE()
Terry Wilson [Thu, 15 Oct 2009 06:48:17 +0000 (06:48 +0000)]
Properly handle PUT requests for CALENDAR_WRITE()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd missing 'getnum' field
Terry Wilson [Wed, 14 Oct 2009 21:16:57 +0000 (21:16 +0000)]
Add missing 'getnum' field

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224074 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAllow for adding message body to the SIP NOTIFY message
Jeff Peeler [Wed, 14 Oct 2009 17:48:57 +0000 (17:48 +0000)]
Allow for adding message body to the SIP NOTIFY message

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

10 years agouse Calendar: instead of Calendar/ for devstate
Terry Wilson [Tue, 13 Oct 2009 22:14:22 +0000 (22:14 +0000)]
use Calendar: instead of Calendar/ for devstate

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix some doxygen format problems and trim trailing whitespace.
Richard Mudgett [Tue, 13 Oct 2009 17:11:46 +0000 (17:11 +0000)]
Fix some doxygen format problems and trim trailing whitespace.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix compiler warning.
Richard Mudgett [Tue, 13 Oct 2009 17:11:05 +0000 (17:11 +0000)]
Fix compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRevert inadvertant code commit to app_originate
Terry Wilson [Tue, 13 Oct 2009 01:58:09 +0000 (01:58 +0000)]
Revert inadvertant code commit to app_originate

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223875 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix handling of notification calls w/ the dialing api
Terry Wilson [Tue, 13 Oct 2009 01:51:46 +0000 (01:51 +0000)]
Fix handling of notification calls w/ the dialing api

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 223804 via svnmerge from
Jeff Peeler [Mon, 12 Oct 2009 23:48:09 +0000 (23:48 +0000)]
Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines

  Ensure ringing continues for branched calls after progress is received

  While waiting for an answer, don't send progress for branched calls
  for which ringing was sent.

  (closes issue #15028)
  Reported by: fnordian
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoClarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options
David Vossel [Mon, 12 Oct 2009 20:58:27 +0000 (20:58 +0000)]
Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options

SWP-151

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223756 65c4cc65-6c06-0410-ace0-fbb531ad65f3