asterisk/asterisk.git
9 years agoLoad all lines from realtime, not just the first one.
Tilghman Lesher [Wed, 23 Jun 2010 18:25:54 +0000 (18:25 +0000)]
Load all lines from realtime, not just the first one.

(closes issue #17144)
 Reported by: nahuelgreco
 Patches:
       20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272145 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake sure reload updates SLA config
Terry Wilson [Wed, 23 Jun 2010 17:21:40 +0000 (17:21 +0000)]
Make sure reload updates SLA config

Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.

(closes issue #16818)
Reported by: mbonin
Patches:
      sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd extra protection for reinvite glare scenario.
Mark Michelson [Wed, 23 Jun 2010 17:08:34 +0000 (17:08 +0000)]
Add extra protection for reinvite glare scenario.

Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.

The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.

The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't try to lock/unlock an uninitialized lock on a dahdi_pri.
Russell Bryant [Tue, 22 Jun 2010 23:20:37 +0000 (23:20 +0000)]
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.

This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes issue with 'dialplan remove extension blah' segfaulting with tab completion
David Vossel [Tue, 22 Jun 2010 22:11:50 +0000 (22:11 +0000)]
fixes issue with 'dialplan remove extension blah' segfaulting with tab completion

(closes issue #17440)
Reported by: kobaz

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoignore CANCEL request after having already received final response to INVITE
David Vossel [Tue, 22 Jun 2010 20:37:05 +0000 (20:37 +0000)]
ignore CANCEL request after having already received final response to INVITE

RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response.  This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".

Review: https://reviewboard.asterisk.org/r/697/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agominor fixes for white/black event filters
David Vossel [Tue, 22 Jun 2010 17:57:28 +0000 (17:57 +0000)]
minor fixes for white/black event filters

This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 271902 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 17:35:17 +0000 (17:35 +0000)]
Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines

  Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.  This is necessary to keep the ref count correct.

  (closes issue #16815)
  Reported by: rain
  Patches:
        chan_sip-unref-fix.diff uploaded by rain (license 327) (modified)
  Tested by: rain
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd regular expression filtering for manager events.
Jeff Peeler [Tue, 22 Jun 2010 16:29:18 +0000 (16:29 +0000)]
Add regular expression filtering for manager events.

This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.

(closes issue #14861)
Reported by: fnordian
Patches:
      eventfilter3.patch uploaded by fnordian (license 110),
      modified by me

Review: https://reviewboard.asterisk.org/r/673/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoResolve some errors that occur on a graceful shutdown.
Russell Bryant [Tue, 22 Jun 2010 16:28:03 +0000 (16:28 +0000)]
Resolve some errors that occur on a graceful shutdown.

Don't Finalize() if Initialize() did not succeed.  This resulted in an error
about trying to Finalize() an invalid handle.

Also trim some trailing whitespace while in the area.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoChange the method of retrieving the Asterisk version string.
Russell Bryant [Tue, 22 Jun 2010 16:17:14 +0000 (16:17 +0000)]
Change the method of retrieving the Asterisk version string.

Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes attended transfer behavior when both transferee and transferer hung up
David Vossel [Tue, 22 Jun 2010 15:46:22 +0000 (15:46 +0000)]
fixes attended transfer behavior when both transferee and transferer hung up

If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to.  This mirrors the expected behavior used in 1.4.

(closes issue #17444)
Reported by: corruptor

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdated the CHANGES file documenting the addition of a configurable port in the dundi...
Matthew Nicholson [Tue, 22 Jun 2010 15:08:39 +0000 (15:08 +0000)]
Updated the CHANGES file documenting the addition of a configurable port in the dundi config file.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 271761 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 14:54:58 +0000 (14:54 +0000)]
Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines

  Allow users to specify a port for dundi peers.

  (closes issue #17056)
  Reported by: klaus3000
  Patches:
        dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
  Tested by: klaus3000
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 271689 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 12:58:28 +0000 (12:58 +0000)]
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines

  Modify chan_sip's packet generation api to automatically calculate the Content-Length.  This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated.  This change was made to ensure that the Content-Length is always correct.

  (closes issue #17326)
  Reported by: kenner
  Tested by: mnicholson, kenner

  Review: https://reviewboard.asterisk.org/r/693/
........

This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoConflict kqueue on OS X, since it doesn't work there yet, anyway.
Tilghman Lesher [Mon, 21 Jun 2010 22:41:00 +0000 (22:41 +0000)]
Conflict kqueue on OS X, since it doesn't work there yet, anyway.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoadd speex 16khz sample frame so codec cost can be calculated
David Vossel [Mon, 21 Jun 2010 21:58:33 +0000 (21:58 +0000)]
add speex 16khz sample frame so codec cost can be calculated

(closes issue #17534)
Reported by: fabled
Patches:
      speex-wb-sample.diff uploaded by fabled (license 448)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 271552 via svnmerge from
Jeff Peeler [Mon, 21 Jun 2010 20:46:53 +0000 (20:46 +0000)]
Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines

  Do not use sizeof to calculate size of a heap allocated character array.

  Change left out from 271399.

  (closes issue #16053)
  Reported by: diLLec
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes crash when From header URI is missing "sip:"
David Vossel [Mon, 21 Jun 2010 20:46:22 +0000 (20:46 +0000)]
fixes crash when From header URI is missing "sip:"

(closes issue #17437)
Reported by: klaus3000
Patches:
      sip_crash uploaded by dvossel (license 671)
Tested by: klaus3000

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes logic error introduced by slin16 sip support
David Vossel [Mon, 21 Jun 2010 20:33:41 +0000 (20:33 +0000)]
fixes logic error introduced by slin16 sip support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd new application for declining counting words in multiple languages.
Tilghman Lesher [Mon, 21 Jun 2010 05:10:06 +0000 (05:10 +0000)]
Add new application for declining counting words in multiple languages.

(closes issue #16869)
 Reported by: chappell
 Patches:
       app_say_counted-20100317.c uploaded by chappell (license 8)
 Tested by: chappell

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 271399 via svnmerge from
Jeff Peeler [Fri, 18 Jun 2010 21:32:09 +0000 (21:32 +0000)]
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines

  Fix crash when parsing some heavily nested statements in AEL on reload.

  Due to the recursion used when compiling AEL in gen_prios, all the stack space
  was being consumed when parsing some AEL that contained nesting 13 levels deep.
  Changing a few large buffers to be heap allocated fixed the crash, although I
  did not test how many more levels can now be safely used.

  (closes issue #16053)
  Reported by: diLLec
  Tested by: jpeeler
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofile.c was truncating audio file formats to the lower 32bits.
David Vossel [Fri, 18 Jun 2010 18:59:05 +0000 (18:59 +0000)]
file.c was truncating audio file formats to the lower 32bits.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRecorded merge of revisions 271335 via svnmerge from
Jeff Peeler [Fri, 18 Jun 2010 18:36:55 +0000 (18:36 +0000)]
Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines

  Eliminate deadlock potential in dahdi_fixup().

  (This is a backport of 269307, committed to trunk by rmudgett.)

  Calling dahdi_indicate() when the channel private lock is already
  held can cause a deadlock if the PRI lock is needed because
  dahdi_indicate() will also get the channel private lock.  The pri_grab()
  function assumes that the channel private lock is held once to avoid
  deadlock.

  (closes issue #17261)
  Reported by: aragon
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271336 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes some coding guideline issue
David Vossel [Thu, 17 Jun 2010 21:23:41 +0000 (21:23 +0000)]
fixes some coding guideline issue

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoretransmit response to BYE requests until timer J expires
David Vossel [Thu, 17 Jun 2010 18:45:32 +0000 (18:45 +0000)]
retransmit response to BYE requests until timer J expires

According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires.  For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms.  This results in some odd behavior.  For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.

To resolve this I made a function called sip_scheddestroy_final().  This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards.  It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time.  This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.

Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy.  sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.

Review: https://reviewboard.asterisk.org/r/694/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoadds support for slin16 in sip
David Vossel [Thu, 17 Jun 2010 18:36:06 +0000 (18:36 +0000)]
adds support for slin16 in sip

(closes issue #16153)
Reported by: kfister
Patches:
      16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
      slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoadds speex 16khz audio support
David Vossel [Thu, 17 Jun 2010 17:23:43 +0000 (17:23 +0000)]
adds speex 16khz audio support

(closes issue #17501)
Reported by: fabled
Patches:
      asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448)
Tested by: malcolmd, fabled, dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoChange expected operation from error to debug message
Jeff Peeler [Thu, 17 Jun 2010 15:34:08 +0000 (15:34 +0000)]
Change expected operation from error to debug message

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 271123 via svnmerge
Matthew Nicholson [Thu, 17 Jun 2010 15:11:55 +0000 (15:11 +0000)]
Blocked revisions 271123 via svnmerge

........
  r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines

  Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct.

  (closes issue #15827)
  Reported by: DennisD
  Patches:
        (modified) dnsmgr_15827.patch uploaded by chappell (license 8)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271124 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agooption w[(secs)] incorrectly capitalized in xmldoc
Paul Belanger [Thu, 17 Jun 2010 00:30:51 +0000 (00:30 +0000)]
option w[(secs)] incorrectly capitalized in xmldoc

(closes issue #17516)
Reported by: karlfife

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoaddition of more parse_uri test cases
David Vossel [Wed, 16 Jun 2010 22:37:45 +0000 (22:37 +0000)]
addition of more parse_uri test cases

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270979 via svnmerge from
Paul Belanger [Wed, 16 Jun 2010 21:17:39 +0000 (21:17 +0000)]
Merged revisions 270979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines

  Fixed typo in macro-page

  Reported to #asterisk-dev by a student of jsmith.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix the actual place that was pointed out, for previous commit.
Jason Parker [Wed, 16 Jun 2010 21:12:25 +0000 (21:12 +0000)]
Fix the actual place that was pointed out, for previous commit.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270983 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270980 via svnmerge from
Jason Parker [Wed, 16 Jun 2010 21:10:48 +0000 (21:10 +0000)]
Merged revisions 270980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines

  Need to lock the agent chan before access its internal bits.

  Pointed out by russellb on asterisk-dev mailing list.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270981 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSet sin_family to AF_INET when doing lookups, also reset sin_port the first time...
Matthew Nicholson [Wed, 16 Jun 2010 20:34:31 +0000 (20:34 +0000)]
Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes.

(closes issue #17496)
Reported by: ManChicken

(closes issue #15827)
Reported by: DennisD
Patches:
      dnsmgr_15827.patch uploaded by chappell (license 8)
Tested by: DennisD, gentlec, damage, wimpy

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoaddition of G.719 pass-through support
David Vossel [Wed, 16 Jun 2010 19:03:24 +0000 (19:03 +0000)]
addition of G.719 pass-through support

(closes issue #16293)
Reported by: malcolmd
Patches:
      g719.passthrough.patch.7 uploaded by malcolmd (license 924)
      format_g719.c uploaded by malcolmd (license 924)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMSG_OOB flag on HANGUP packet removed.
Paul Belanger [Wed, 16 Jun 2010 18:43:22 +0000 (18:43 +0000)]
MSG_OOB flag on HANGUP packet removed.

Per Tilghman's request on IRC (#asterisk-bugs).

(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270866 via svnmerge from
David Vossel [Wed, 16 Jun 2010 17:36:51 +0000 (17:36 +0000)]
Merged revisions 270866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines

  fixes chan_iax2 race condition

  There is code in chan_iax2.c that attempts to guarantee that only a single
  active thread will handle a call number at a time.  This code works once
  the thread is added to an active_list of threads, but we are not currently
  guaranteed that a newly activated thread will enter the active_list immediately
  because it is left up to the thread to add itself after frames have been
  queued to it.  This means that if two frames come in for the same call number
  at the same time, it is possible for them to grab two separate threads because
  the first thread did not add itself to the active_list fast enough.  This
  causes some pretty complex problems.

  This patch resolves this race condition by immediately adding an activated
  thread to the active_list within the network thread and only depending on
  the thread to remove itself once it is done processing the frames queued to
  it.  By doing this we are guaranteed that if another frame for the same call
  number comes in at the same time, that this thread will immediately be found
  in the active_list of threads.

  Review: https://reviewboard.asterisk.org/r/720/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix no call waiting caller ID
Jeff Peeler [Wed, 16 Jun 2010 16:45:07 +0000 (16:45 +0000)]
Fix no call waiting caller ID

Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdate formatting for channelvariables.tex
Paul Belanger [Wed, 16 Jun 2010 15:05:11 +0000 (15:05 +0000)]
Update formatting for channelvariables.tex

(closes issue #17511)
Reported by: klaus3000
Patches:
      channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't blow up if an ast_channel doesn't get allocated.
Russell Bryant [Tue, 15 Jun 2010 22:48:12 +0000 (22:48 +0000)]
Don't blow up if an ast_channel doesn't get allocated.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't continue sending the file when there has been an error
Terry Wilson [Tue, 15 Jun 2010 21:42:33 +0000 (21:42 +0000)]
Don't continue sending the file when there has been an error

If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.

(closes issue #16682)
Reported by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270692 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't send files twice and remove extra \r\n from header
Terry Wilson [Tue, 15 Jun 2010 21:10:15 +0000 (21:10 +0000)]
Don't send files twice and remove extra \r\n from header

After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.

(closes issue #17239)
Reported by: cjacobsen
Patches:
      patch2.diff uploaded by cjacobsen (license 1029)
Tested by: lathama, cjacobsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270660 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake contactdeny apply to src ip when nat=yes
Terry Wilson [Tue, 15 Jun 2010 20:18:04 +0000 (20:18 +0000)]
Make contactdeny apply to src ip when nat=yes

chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.

Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.

This commit makes contactdeny apply to the src ip when nat=yes instead.

(closes issue #17276)
Reported by: klaus3000
Patches:
      patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270583 via svnmerge from
Tilghman Lesher [Tue, 15 Jun 2010 18:26:26 +0000 (18:26 +0000)]
Merged revisions 270583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines

  Variables have always been case-sensitive, so we should not be removing case-insensitive matches.

  Bug reported via the -dev list.  See
  http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoArgh, mixed declarations and code.
Tilghman Lesher [Tue, 15 Jun 2010 18:16:04 +0000 (18:16 +0000)]
Argh, mixed declarations and code.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd distributed devicestate via the XMPP protocol.
Tilghman Lesher [Tue, 15 Jun 2010 17:06:23 +0000 (17:06 +0000)]
Add distributed devicestate via the XMPP protocol.

(closes issue #15757)
 Reported by: Marquis
 Patches:
       distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
 Tested by: Marquis, lmadsen, marcelloceschia

Review: https://reviewboard.asterisk.org/r/351/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270442 via svnmerge from
Leif Madsen [Tue, 15 Jun 2010 12:51:37 +0000 (12:51 +0000)]
Merged revisions 270442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line

  Move information about zonemessages into the [zonemessages] section.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270443 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270331 via svnmerge from
Paul Belanger [Mon, 14 Jun 2010 21:33:55 +0000 (21:33 +0000)]
Merged revisions 270331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines

  Properly play first file in sort list.

  When using sort=alpha we would always skip the first file
  in the list first time through.  We now check for that
  properly.

  (closes issue #17470)
  Reported by: pabelanger
  Patches:
        sort.aplha.patch uploaded by pabelanger (license 224)
  Tested by: lmadsen

  Review: https://reviewboard.asterisk.org/r/703/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270332 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoExtract sig_ss7_init_linkset() to sig_ss7.
Richard Mudgett [Mon, 14 Jun 2010 20:51:09 +0000 (20:51 +0000)]
Extract sig_ss7_init_linkset() to sig_ss7.

Also found a place where sig_pri_init_pri() was inlined and called it
instead.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd option to get untruncated channel name from AGENT function.
Jason Parker [Mon, 14 Jun 2010 19:41:43 +0000 (19:41 +0000)]
Add option to get untruncated channel name from AGENT function.

The "channel" option would chop the channel name at the last '-', which made
it useless for something like a channel transfer from the dialplan.  The
"fullchannel" option will return the channel name as-is.

ABE-2218

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
Richard Mudgett [Mon, 14 Jun 2010 15:55:35 +0000 (15:55 +0000)]
Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.

Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn.

Review: https://reviewboard.asterisk.org/r/696/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agobashism in configure script
Tzafrir Cohen [Sun, 13 Jun 2010 09:16:25 +0000 (09:16 +0000)]
bashism in configure script

Theoretically the ./configure script is a pure bourne-shell script.
Practically it may be run by bash if /bin/sh is not good enough. But we should not count on it. See bug report for the gory details.

(closes issue #17485)
Patches:
      0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by tzafrir (license 46)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReverting patch and reopening issue #16155, as patch breaks
Paul Belanger [Sun, 13 Jun 2010 01:53:54 +0000 (01:53 +0000)]
Reverting patch and reopening issue #16155, as patch breaks
FreeBSD / OSX builds.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 270078 via svnmerge from
Paul Belanger [Sat, 12 Jun 2010 18:55:47 +0000 (18:55 +0000)]
Merged revisions 270078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines

  Fix typo in example
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse pkg-config to find gmime libraries
Paul Belanger [Fri, 11 Jun 2010 20:14:13 +0000 (20:14 +0000)]
Use pkg-config to find gmime libraries

This way the libraries can be found even if they are in
non-standard locations.

(closes issue #16155)
Reported by: jcollie
Patches:
      0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412)
Tested by: jsmith, tilghman, pabelanger

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269960 via svnmerge from
Tilghman Lesher [Fri, 11 Jun 2010 18:31:14 +0000 (18:31 +0000)]
Merged revisions 269960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines

  For SpeeX, 0 bits remaining is valid and does not need an emitted warning.

  (closes issue #15762)
   Reported by: nblasgen
   Patches:
         issue15672.patch uploaded by pabelanger (license 224)
   Tested by: nblasgen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd DBGetComplete event after a DBGetResponse.
Tilghman Lesher [Fri, 11 Jun 2010 18:17:28 +0000 (18:17 +0000)]
Add DBGetComplete event after a DBGetResponse.

(closes issue #16965)
 Reported by: rrb3942
 Patches:
       DBGetComplete.patch uploaded by rrb3942 (license 1003)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269938 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove lines from the output related to the backtrace itself.
Tilghman Lesher [Fri, 11 Jun 2010 18:04:54 +0000 (18:04 +0000)]
Remove lines from the output related to the backtrace itself.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove ASTBINDIR variable
Paul Belanger [Thu, 10 Jun 2010 20:30:44 +0000 (20:30 +0000)]
Remove ASTBINDIR variable

(closes issue #17031)
Reported by: pabelanger
Patches:
      Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269821 via svnmerge from
Mark Michelson [Thu, 10 Jun 2010 19:34:03 +0000 (19:34 +0000)]
Merged revisions 269821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines

  Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.

  The issue here was that the frame created when adjusting for PLC had no offset
  to its audio data. If this frame were translated to another format prior to
  being sent out an RTP socket, all went well because the translation code would
  put an appropriate offset into the frame. However, if the SLIN audio were not
  translated before being sent out the RTP socket, bad things would happen.
  Specifically, the ast_rtp_raw_write makes the assumption that the frame has
  at least enough of an offset that it can accommodate an RTP header. This was
  not the case. As such, data was being written prior to the allocation, likely
  corrupting the data the memory allocator had written. Thus when the time came
  to free the data, all hell broke loose. ....Well, Asterisk crashed at least.

  The fix was just what one would expect. Offset the data in the frame by a reasonable
  amount. The method I used is a bit odd since the data in the frame is 16 bit integers
  and not bytes. I left a big ol' comment about it. This can be improved on if someone
  is interested. I was more interested in getting the crash resolved.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd documentation explaining PLC in Asterisk.
Mark Michelson [Thu, 10 Jun 2010 17:14:38 +0000 (17:14 +0000)]
Add documentation explaining PLC in Asterisk.

Review: https://reviewboard.asterisk.org/r/688/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix an off by one error that caused a unit test to occasionally crash.
Russell Bryant [Thu, 10 Jun 2010 13:17:51 +0000 (13:17 +0000)]
Fix an off by one error that caused a unit test to occasionally crash.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEnsure that 'logger show channels' works properly when wildcards are used in logger...
Kevin P. Fleming [Thu, 10 Jun 2010 12:28:17 +0000 (12:28 +0000)]
Ensure that 'logger show channels' works properly when wildcards are used in logger.conf.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269635 via svnmerge from
Tilghman Lesher [Thu, 10 Jun 2010 08:15:45 +0000 (08:15 +0000)]
Merged revisions 269635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines

  Ensure restartable system calls can restart (BSD signal semantics).

  This eliminates the annoying <beep> on the console.

  (closes issue #17477)
   Reported by: jvandal
   Patches:
         20100610__issue17477.diff.txt uploaded by tilghman (license 14)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAttempt to fix a FreeBSD build error by including sys/stat.h.
Russell Bryant [Thu, 10 Jun 2010 00:32:31 +0000 (00:32 +0000)]
Attempt to fix a FreeBSD build error by including sys/stat.h.

http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269602 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAttempt to fix FreeBSD build problem.
Russell Bryant [Wed, 9 Jun 2010 23:56:08 +0000 (23:56 +0000)]
Attempt to fix FreeBSD build problem.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269495 via svnmerge from
Russell Bryant [Wed, 9 Jun 2010 22:19:20 +0000 (22:19 +0000)]
Merged revisions 269495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines

  Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 269426 via svnmerge
Jason Parker [Wed, 9 Jun 2010 21:38:33 +0000 (21:38 +0000)]
Blocked revisions 269426 via svnmerge

........
  r269426 | qwell | 2010-06-09 16:19:17 -0500 (Wed, 09 Jun 2010) | 6 lines

  Let systems without a working fork() use res_musiconhold.

  Files mode doesn't require anything special, so that can still be used just fine.

  AST-357
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269486 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoResolve an invalid memory read on an event.
Russell Bryant [Wed, 9 Jun 2010 21:11:43 +0000 (21:11 +0000)]
Resolve an invalid memory read on an event.

Valgrind pointed out that attempting to get an IE value from an event that has
no IEs produces an invalid memory read past the end of the event.  Thanks to
mmichelson for pointing the problem out to me and then testing the fix.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269334 via svnmerge from
Paul Belanger [Wed, 9 Jun 2010 17:32:52 +0000 (17:32 +0000)]
Merged revisions 269334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines

  Fix Debian init script to not use -c.

  When using the init script as-is currently, it could cause issues on Debian
  such as high CPU usage. This fix has worked for several people so I'm
  implementing the change.  We now handle color displays properly.

  (closes issue #16784)
  Reported by: pabelanger
  Patches:
        20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
  Tested by: pabelanger, tilghman
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269346 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd missing API function to sig_ss7: sig_ss7_fixup().
Richard Mudgett [Wed, 9 Jun 2010 17:06:41 +0000 (17:06 +0000)]
Add missing API function to sig_ss7: sig_ss7_fixup().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269308 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEliminate deadlock potential in dahdi_fixup().
Richard Mudgett [Wed, 9 Jun 2010 16:54:38 +0000 (16:54 +0000)]
Eliminate deadlock potential in dahdi_fixup().

Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are
in a potentially inconsistent state is a potentially bad thing in
principle.

However, calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock.  The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes crash in moh when cachertclasses flag is used
David Vossel [Wed, 9 Jun 2010 15:09:25 +0000 (15:09 +0000)]
fixes crash in moh when cachertclasses flag is used

The result for moh_register was not verified to guarantee
the mohclass as added to the container.

(closes issue #16993)
Reported by: dmitri
Patches:
      res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
      moh_crash2.diff uploaded by dvossel (license 671)
Tested by: dmitri

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269271 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agodial by name in chan_dahdi
Tzafrir Cohen [Wed, 9 Jun 2010 13:17:43 +0000 (13:17 +0000)]
dial by name in chan_dahdi

* chan_dahdi supports dialing configuring and dialing by device file name.
  DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
  it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
  False by default. If set, chan_dahdi will ignore failed 'channel' entries.
  Handy for the above name-based syntax as it does not depend on
  initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
  (gGrR) dialing, which make it lsightly more complicated.

https://reviewboard.asterisk.org/r/535/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libjack-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:55:07 +0000 (10:55 +0000)]
Add libjack-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libpopt-dev, libical-dev, and libspandsp-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:53:26 +0000 (10:53 +0000)]
Add libpopt-dev, libical-dev, and libspandsp-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269204 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libnewt-dev to install-prereq.
Russell Bryant [Wed, 9 Jun 2010 10:48:29 +0000 (10:48 +0000)]
Add libnewt-dev to install-prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libopenais-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:47:19 +0000 (10:47 +0000)]
Add libopenais-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269202 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd an "install-unpackaged" command to install_prereq for installing unpackaged depen...
Russell Bryant [Wed, 9 Jun 2010 10:45:10 +0000 (10:45 +0000)]
Add an "install-unpackaged" command to install_prereq for installing unpackaged dependencies (such as NBS and libresample).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libcurl to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:33:32 +0000 (10:33 +0000)]
Add libcurl to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269200 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd freetds-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:30:32 +0000 (10:30 +0000)]
Add freetds-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269199 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libradiusclient-ng-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:28:27 +0000 (10:28 +0000)]
Add libradiusclient-ng-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269198 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libbluetooth-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:23:05 +0000 (10:23 +0000)]
Add libbluetooth-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libmysqlclient-dev to install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:21:23 +0000 (10:21 +0000)]
Add libmysqlclient-dev to install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd libgtk2.0-dev to the packages list for install_prereq.
Russell Bryant [Wed, 9 Jun 2010 10:18:24 +0000 (10:18 +0000)]
Add libgtk2.0-dev to the packages list for install_prereq.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269187 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd High Resolution Times to CDRs for Asterisk
Bradley Latus [Tue, 8 Jun 2010 23:48:17 +0000 (23:48 +0000)]
Add High Resolution Times to CDRs for Asterisk

People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.

Patch by snuffy.

(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy

Review: https://reviewboard.asterisk.org/r/461/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix build on Mac OS X (and maybe FreeBSD, too)
Tilghman Lesher [Tue, 8 Jun 2010 22:45:16 +0000 (22:45 +0000)]
Fix build on Mac OS X (and maybe FreeBSD, too)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269119 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't pass null to manager_event()
Matthew Nicholson [Tue, 8 Jun 2010 18:50:45 +0000 (18:50 +0000)]
Don't pass null to manager_event()

(closes issue #17087)
Reported by: bklang
Patches:
      app-fax-null-sprintf1.diff uploaded by mnicholson (license 96)
Tested by: bklang

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269083 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEnsure CONFIG_FLAGS makes it into the build rules when doing out of tree builds.
Russell Bryant [Tue, 8 Jun 2010 15:41:23 +0000 (15:41 +0000)]
Ensure CONFIG_FLAGS makes it into the build rules when doing out of tree builds.

(closes issue #16685)
Reported by: pprindeville

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 269006 via svnmerge from
Sean Bright [Tue, 8 Jun 2010 15:39:52 +0000 (15:39 +0000)]
Merged revisions 269006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun 2010) | 11 lines

  Reduce startup time for cdr_tds with large CDR tables.

  Since we are just checking for table existence, add a WHERE clause that will
  return no rows but will raise an error if the table doesn't exist.

  (closes issue #17380)
  Reported by: kkwong
  Patches:
        issue17380-01.patch uploaded by seanbright (license 71)
  Tested by: kkwong
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdate note in sip.conf.sample.
Leif Madsen [Tue, 8 Jun 2010 15:23:20 +0000 (15:23 +0000)]
Update note in sip.conf.sample.
Update note in sip.conf.sample about externip and externhost with STUN.

(closes issue #16323)
Reported by: klaus3000
Patches:
      sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix some doxygen warnings.
Leif Madsen [Tue, 8 Jun 2010 14:38:18 +0000 (14:38 +0000)]
Fix some doxygen warnings.

(closes issue #17336)
Reported by: snuffy
Patches:
      doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRelease list lock before returning on error.
Tilghman Lesher [Tue, 8 Jun 2010 06:57:24 +0000 (06:57 +0000)]
Release list lock before returning on error.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268933 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix trunk build on Mac OS X.
Tilghman Lesher [Tue, 8 Jun 2010 06:16:43 +0000 (06:16 +0000)]
Fix trunk build on Mac OS X.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd SRTP support for Asterisk
Terry Wilson [Tue, 8 Jun 2010 05:29:08 +0000 (05:29 +0000)]
Add SRTP support for Asterisk

After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake SIP tests compile again.
Richard Mudgett [Tue, 8 Jun 2010 00:45:13 +0000 (00:45 +0000)]
Make SIP tests compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse the mailbox destructor function, instead.
Tilghman Lesher [Mon, 7 Jun 2010 22:56:53 +0000 (22:56 +0000)]
Use the mailbox destructor function, instead.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMailbox list would previously grow at each reload, containing duplicates.
Tilghman Lesher [Mon, 7 Jun 2010 22:47:13 +0000 (22:47 +0000)]
Mailbox list would previously grow at each reload, containing duplicates.

Also, optimize the allocation of mailboxes to avoid additional memory structures.

(closes issue #16320)
 Reported by: Marquis
 Patches:
       20100525__issue16320.diff.txt uploaded by tilghman (license 14)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268817 65c4cc65-6c06-0410-ace0-fbb531ad65f3