asterisk/asterisk.git
10 years agoMerged revisions 209759 via svnmerge from
Kevin P. Fleming [Sat, 1 Aug 2009 01:03:07 +0000 (01:03 +0000)]
Merged revisions 209759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines

  Minor changes inspired by testing with latest GCC.

  The latest GCC (what will become 4.5.x) has a few new warnings, that in these
  cases found some either downright buggy code, or at least seriously poorly
  designed code that could be improved.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix some places where ast_event_type was used instead of ast_event_ie_type.
Russell Bryant [Fri, 31 Jul 2009 21:53:31 +0000 (21:53 +0000)]
Fix some places where ast_event_type was used instead of ast_event_ie_type.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd configuration sample code for previous commit.
Mark Michelson [Fri, 31 Jul 2009 17:57:00 +0000 (17:57 +0000)]
Add configuration sample code for previous commit.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209674 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoImprove chan_sip's ability to determine what methods should and should not be used...
Mark Michelson [Fri, 31 Jul 2009 17:55:44 +0000 (17:55 +0000)]
Improve chan_sip's ability to determine what methods should and should not be used in a dialog.

The previous effort here was to store what a peer is capable of receiving by parsing REGISTER
requests from the peer and keeping that information for as long as the registration was active.
The problem with this is that there are a great number of SIP devices which give no indication
of the methods allowed in their REGISTER requests, and it is unreasonable to try to guess what
the device may or may not support. In addition, some SIP devices have been found to claim support
for a specific method, but their handling the method is less than ideal, or they are actually
lying.

With this patch, we now determine what methods a device supports  by parsing the Allow header we
receive from them, and we do this with each new dialog. In addition, a configuration option has
been added so that an administrator can essentially blacklist certain methods from being used
with certain peers if the admin knows that support for a specific method is dodgy or nonexistent.

ABE-1822

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAllow passing 'noisy' to configure's --enable-dev-mode argument to turn on verbose...
Sean Bright [Thu, 30 Jul 2009 23:37:31 +0000 (23:37 +0000)]
Allow passing 'noisy' to configure's --enable-dev-mode argument to turn on verbose builds.

(closes issue #15607)
Reported by: mvanbaak
Patches:
      20090730_issue15607.patch uploaded by seanbright (license 71)
Tested by: seanbright

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd missing ifdef-s for service maintenance message functionality
Jeff Peeler [Thu, 30 Jul 2009 23:31:41 +0000 (23:31 +0000)]
Add missing ifdef-s for service maintenance message functionality

(closes issue #15614)
Reported by: fabled

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFixes numerous spelling errors. Patch submitted by alecdavis.
David Brooks [Thu, 30 Jul 2009 16:07:05 +0000 (16:07 +0000)]
Fixes numerous spelling errors. Patch submitted by alecdavis.

(closes issue #15595)
Reported by: alecdavis

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix a crash that can result if text codecs are allowed but textsupport is disabled.
Mark Michelson [Thu, 30 Jul 2009 14:38:21 +0000 (14:38 +0000)]
Fix a crash that can result if text codecs are allowed but textsupport is disabled.

(closes issue #15596)
Reported by: fabled
Patches:
      sip-red.patch uploaded by fabled (license 448)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209516 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoThis patch adds the ability to send a CUSD command to a bluetooth device.
Matthew Nicholson [Wed, 29 Jul 2009 21:46:17 +0000 (21:46 +0000)]
This patch adds the ability to send a CUSD command to a bluetooth device.

(closes issue #15278)
Reported by: Artem
Patches:
      cusd5.patch uploaded by Artem (license 800)
Tested by: mnicholson, Artem

Review: https://reviewboard.asterisk.org/r/274/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFixed a comment for hfp_parse_clip
Matthew Nicholson [Wed, 29 Jul 2009 21:13:42 +0000 (21:13 +0000)]
Fixed a comment for hfp_parse_clip

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209453 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoDefine side-effect-safe MIN and MAX macros and remove duplicate definitions from...
Kevin P. Fleming [Tue, 28 Jul 2009 13:49:46 +0000 (13:49 +0000)]
Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRegex FTL
Tilghman Lesher [Tue, 28 Jul 2009 00:20:26 +0000 (00:20 +0000)]
Regex FTL

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 209315 via svnmerge from
Tilghman Lesher [Tue, 28 Jul 2009 00:14:12 +0000 (00:14 +0000)]
Merged revisions 209315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines

  Publish French extra sounds
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoCleanup T.38 negotiation changes.
Kevin P. Fleming [Mon, 27 Jul 2009 21:43:36 +0000 (21:43 +0000)]
Cleanup T.38 negotiation changes.

Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
clean up some looping logic, and correct an improper use of ast_free() for
freeing an ast_frame.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake T.38 switchover in ReceiveFAX synchronous.
Kevin P. Fleming [Mon, 27 Jul 2009 21:21:43 +0000 (21:21 +0000)]
Make T.38 switchover in ReceiveFAX synchronous.

In receive mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening for an incoming
CNG tone and only triggering on that. The channel may be using a low-bitrate
codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
at all, or there could be a variety of other reasons that we don't detect it,
but in all those cases if T.38 is available we certainly want to use it.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209256 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoGracefully handle malformed RTP text packets.
Mark Michelson [Mon, 27 Jul 2009 20:54:54 +0000 (20:54 +0000)]
Gracefully handle malformed RTP text packets.

AST-2009-004

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209235 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoHonor channel's music class when using realtime music on hold.
Mark Michelson [Mon, 27 Jul 2009 20:11:42 +0000 (20:11 +0000)]
Honor channel's music class when using realtime music on hold.

(closes issue #15051)
Reported by: alexh
Patches:
      15051.patch uploaded by mmichelson (license 60)
Tested by: alexh

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 209131 via svnmerge from
Mark Michelson [Mon, 27 Jul 2009 17:50:04 +0000 (17:50 +0000)]
Merged revisions 209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines

  Allow for UDPTL to use only even-numbered ports if desired.

  There are some VoIP providers out there that will not accept SDP
  offers with odd numbered UDPTL ports. While it is my personal opinion
  that these VoIP providers are misinterpreting RFC 2327, it really is
  not a big deal to play along with their silly little games. Of course,
  since restricting UDPTL ports to only even numbers reduces the range
  of available ports by half, so the option to use only even port numbers
  is off by default. A user can enable the behavior by setting
  use_even_ports=yes in udptl.conf.

  (closes issue #15182)
  Reported by: CGMChris
  Patches:
        15182.patch uploaded by mmichelson (license 60)
  Tested by: CGMChris
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
David Brooks [Mon, 27 Jul 2009 16:33:50 +0000 (16:33 +0000)]
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"

(closes issue #15571)
Reported by: alecdavis

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRestore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes.
Kevin P. Fleming [Mon, 27 Jul 2009 15:38:59 +0000 (15:38 +0000)]
Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes.

During the recent Makefile improvements I made, it seemed the 'make' was
automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes,
so I removed the explict export of them. However, there are some circumstances
where make does this, and some where it does not, so I've brought them back
to ensure they are always exported. I also removed an extraneous double setting
of _ASTLDFLAGS on *BSD platforms.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 208990 via svnmerge
Michiel van Baak [Mon, 27 Jul 2009 09:56:49 +0000 (09:56 +0000)]
Blocked revisions 208990 via svnmerge

........
  r208990 | mvanbaak | 2009-07-27 11:56:13 +0200 (Mon, 27 Jul 2009) | 5 lines

  backport rev 205532 from trunk:

  pthread_self returns a pthread_t which is not an unsigned int on all
  pthread implementations. Casting it to an unsigned int fixes compiler warnings.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208991 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208923 via svnmerge from
Jeff Peeler [Mon, 27 Jul 2009 01:20:37 +0000 (01:20 +0000)]
Merged revisions 208923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines

  Fix logic errors from 208746
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoadd OpenBSD to the install_prereq script
Michiel van Baak [Sun, 26 Jul 2009 14:00:52 +0000 (14:00 +0000)]
add OpenBSD to the install_prereq script

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agolibxml2-dev is needed as well by default.
Michiel van Baak [Sat, 25 Jul 2009 12:28:38 +0000 (12:28 +0000)]
libxml2-dev is needed as well by default.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoadd default alias reload to run module reload.
Michiel van Baak [Sat, 25 Jul 2009 12:03:25 +0000 (12:03 +0000)]
add default alias reload to run module reload.

Requiring 'module reload' to reload everything, including
core etc makes russell very unhappy.

The default configuration already loads the 'friendly' aliases template.
Added 'reload=module reload' to that template.

Also removed the comment in main/cli.c that reload should come back.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208746 via svnmerge from
Jeff Peeler [Sat, 25 Jul 2009 06:23:18 +0000 (06:23 +0000)]
Merged revisions 208746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines

  Fix compiling under dev-mode with gcc 4.4.0.

  Mostly trivial changes, but I did not know of any other way to fix the
  "dereferencing type-punned pointer will break strict-aliasing rules" error
  without creating a tmp variable in chan_skinny.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove trailing whitespace.
Russell Bryant [Fri, 24 Jul 2009 21:12:43 +0000 (21:12 +0000)]
Remove trailing whitespace.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoNote that "reload" needs to be added back.
Russell Bryant [Fri, 24 Jul 2009 20:54:37 +0000 (20:54 +0000)]
Note that "reload" needs to be added back.

I keep getting annoyed at having to type "module reload" to reload everything,
so I'm adding a note that we need to add "reload" back.  "module reload" doesn't
really make sense as the command to reload everything, including the core.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208706 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoDon't log a warning for something that does not affect operation.
Russell Bryant [Fri, 24 Jul 2009 20:25:23 +0000 (20:25 +0000)]
Don't log a warning for something that does not affect operation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 208622 via svnmerge
Mark Michelson [Fri, 24 Jul 2009 19:26:26 +0000 (19:26 +0000)]
Blocked revisions 208622 via svnmerge

........
  r208622 | mmichelson | 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) | 16 lines

  Don't impose an arbitrary limit on member lines in queues.conf

  I know what some of you are thinking: "UGH! Mark, why are you using
  ast_strdup and ast_free for the string when you can just use ast_strdupa
  and let the memory free itself?! Have the bats been chewing on your brain
  again?"

  Based on past experiences, I don't like using ast_strdupa inside a loop.
  It's a good way to potentially exhaust stack space. Also, since this only
  happens when reloading queues, I don't think that heap allocations and
  frees are going to be a huge problem.

  (closes issue #15559)
  Reported by: amorsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208630 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208592 via svnmerge from
Russell Bryant [Fri, 24 Jul 2009 18:42:32 +0000 (18:42 +0000)]
Merged revisions 208592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines

  Do not log an ERROR if autoservice_stop() returns -1.

  This does not indicate an error.  A return of -1 just means that the channel
  has been hung up.

  (reported in #asterisk-dev)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208587 via svnmerge from
Mark Michelson [Fri, 24 Jul 2009 18:31:04 +0000 (18:31 +0000)]
Merged revisions 208587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines

  Only send a BYE when hanging up a channel that is up.

  For cases where Asterisk sends an INVITE and receives a non 2XX final
  response, Asterisk would follow the INVITE transaction by immediately
  sending a BYE, which was unnecessary.

  (closes issue #14575)
  Reported by: chris-mac
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208588 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoResolve a T.38 negotiation issue left over from the udptl-updates merge.
Kevin P. Fleming [Fri, 24 Jul 2009 15:02:53 +0000 (15:02 +0000)]
Resolve a T.38 negotiation issue left over from the udptl-updates merge.

The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agouse aptitude for debian based systems
Michiel van Baak [Fri, 24 Jul 2009 14:35:49 +0000 (14:35 +0000)]
use aptitude for debian based systems

The function to check wether we need to install packages was using
dpkg-query which was gives wrong output on Debian 5

Also, the apt-get has been replaced with aptitude because aptitude
is now the preferred way to handle packages on Debian

(closes issue #15570)
Reported by: mvanbaak
Patches:
      2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoT.38 change note is not necessary in this branch
Kevin P. Fleming [Thu, 23 Jul 2009 22:32:52 +0000 (22:32 +0000)]
T.38 change note is not necessary in this branch

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208504 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRework of T.38 negotiation and UDPTL API to address interoperability problems
Kevin P. Fleming [Thu, 23 Jul 2009 21:57:24 +0000 (21:57 +0000)]
Rework of T.38 negotiation and UDPTL API to address interoperability problems

Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.

The major changes here are:

1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.

2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.

3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.

4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.

Review: https://reviewboard.asterisk.org/r/310/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208386 via svnmerge from
Mark Michelson [Thu, 23 Jul 2009 19:34:49 +0000 (19:34 +0000)]
Merged revisions 208386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines

  Fix a problem where a 491 response could be sent out of dialog.

  This generalizes the fix for issue 13849. The initial fix corrected the
  problem that Asterisk would reply with a 491 if a reinvite were received
  from an endpoint and we had not yet received an ACK from that endpoint
  for the initial INVITE it had sent us. This expansion also allows Asterisk
  to appropriately handle an INVITE with authorization credentials if Asterisk
  had not received an ACK from the previous transaction in which Asterisk had
  responded to an unauthorized INVITE with a 407.

  (closes issue #14239)
  Reported by: klaus3000
  Patches:
        14239.patch uploaded by mmichelson (license 60)
  Tested by: klaus3000

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208388 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208380 via svnmerge from
Jeff Peeler [Thu, 23 Jul 2009 19:21:50 +0000 (19:21 +0000)]
Merged revisions 208380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines

  Only set the priindication setting when not performing a reload

  (closes issue #14696)
  Reported by: fdecher
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208383 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208312 via svnmerge from
Mark Michelson [Thu, 23 Jul 2009 16:29:37 +0000 (16:29 +0000)]
Merged revisions 208312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines

  Remove inaccurate XXX comment.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix sending of interface identifier unconditionally in sig_pri
Jeff Peeler [Thu, 23 Jul 2009 15:59:44 +0000 (15:59 +0000)]
Fix sending of interface identifier unconditionally in sig_pri

The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.

(closes issue #15452)
Reported by: alecdavis
Patches:
      bug15452.patch uploaded by jpeeler (license 325)
Tested by: alecdavis

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208267 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208262 via svnmerge from
Mark Michelson [Thu, 23 Jul 2009 15:46:34 +0000 (15:46 +0000)]
Merged revisions 208262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines

  Properly handle 183 responses which do not contain an SDP.

  (closes issue #15442)
  Reported by: ffloimair
  Patches:
        15442.patch uploaded by mmichelson (license 60)
  Tested by: tkarl, ffloimair
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix potential crash if p->owner is NULL.
Mark Michelson [Thu, 23 Jul 2009 14:46:53 +0000 (14:46 +0000)]
Fix potential crash if p->owner is NULL.

Problem was observed when a call-forwarding loop was accidentally
configured.

ABE-1906

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoResolve compiler warning on mac.
Russell Bryant [Thu, 23 Jul 2009 01:31:18 +0000 (01:31 +0000)]
Resolve compiler warning on mac.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoReset the fax buffers back to default settings regardless of signaling in use -
Jeff Peeler [Wed, 22 Jul 2009 22:42:33 +0000 (22:42 +0000)]
Reset the fax buffers back to default settings regardless of signaling in use -
Pointed out by Matt F.
Also in the case of not using a signaling module, set the law back to the
default as well.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 208083 via svnmerge from
Tilghman Lesher [Wed, 22 Jul 2009 22:35:57 +0000 (22:35 +0000)]
Merged revisions 208083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) | 4 lines

  Export symbols for functions included in our compatibility headers.
  (closes issue #15556)
   Reported by: smw1218
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRestore an int declaration on PPC platforms.
Jason Parker [Wed, 22 Jul 2009 21:43:57 +0000 (21:43 +0000)]
Restore an int declaration on PPC platforms.

This x is one crafty little bugger...
It was used for 2 different things (one of which was only done on PPC) in 1.4.
One of the uses were removed in trunk, and with it went the declaration.

(closes issue #14038)
Reported by: ffloimair

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208113 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoClarify documentation on 'realtime update2' to show more than one condition.
Tilghman Lesher [Wed, 22 Jul 2009 16:49:42 +0000 (16:49 +0000)]
Clarify documentation on 'realtime update2' to show more than one condition.
(closes issue #15357)
 Reported by: snuffy
 Patches:
       bug_fix_doc_update2.diff uploaded by snuffy (license 35)
       (slightly modified by me)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove trailing whitespace.
Russell Bryant [Wed, 22 Jul 2009 14:35:49 +0000 (14:35 +0000)]
Remove trailing whitespace.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix the crash in directed pickups. For real this time.
Mark Michelson [Wed, 22 Jul 2009 14:35:01 +0000 (14:35 +0000)]
Fix the crash in directed pickups. For real this time.

A shallow pointer copy was causing an ast_party_connected_line
structure to be freed multiple times, thus causing a crash.

(closes issue #15441)
Reported by: lmsteffan
Patches:
      15441.patch uploaded by mmichelson (license 60)
Tested by: lmsteffan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoDo not dial digits when none were specified for sig_pri based calls
Jeff Peeler [Tue, 21 Jul 2009 22:51:47 +0000 (22:51 +0000)]
Do not dial digits when none were specified for sig_pri based calls

(closes issue #15524)
Reported by: elguero
Patches:
      pri-sig-no-dest-set.patch uploaded by elguero (license 37)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207945 via svnmerge from
Tilghman Lesher [Tue, 21 Jul 2009 22:45:32 +0000 (22:45 +0000)]
Merged revisions 207945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines

  Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
  This change makes URIENCODE and QUOTE behave similarly, since the documentation
  states that the argument is not optional, for both.
  (closes issue #15439)
   Reported by: pkempgen
   Patches:
         20090706__issue15439.diff.txt uploaded by tilghman (license 14)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agowhitespace fix only
Jeff Peeler [Tue, 21 Jul 2009 22:24:56 +0000 (22:24 +0000)]
whitespace fix only

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207934 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoNote that we use tabs instead of spaces for indentation.
Russell Bryant [Tue, 21 Jul 2009 22:22:18 +0000 (22:22 +0000)]
Note that we use tabs instead of spaces for indentation.

I'm surprised this was never actually in here...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207925 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix my_is_off_hook to check rxbits only for FXS signaling
Jeff Peeler [Tue, 21 Jul 2009 22:02:25 +0000 (22:02 +0000)]
Fix my_is_off_hook to check rxbits only for FXS signaling

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207902 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207827 via svnmerge from
Jeff Peeler [Tue, 21 Jul 2009 20:26:02 +0000 (20:26 +0000)]
Merged revisions 207827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines

  Wait for wink before dialing when using E&M wink signaling

  There was already code for other signaling types in dahdi_handle_event to
  handle dialing if a dial operation dial string was present. Simply add
  SIG_EMWINK to the list.

  (closes issue #14434)
  Reported by: araasch
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207854 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207714 via svnmerge from
Mark Michelson [Tue, 21 Jul 2009 14:29:40 +0000 (14:29 +0000)]
Merged revisions 207714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines

  Document default timeout for AMI originations.

  AST-224
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207723 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207647 via svnmerge from
Kevin P. Fleming [Tue, 21 Jul 2009 13:28:04 +0000 (13:28 +0000)]
Merged revisions 207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines

  Ensure that user-provided CFLAGS and LDFLAGS are honored.

  This commit changes the build system so that user-provided flags (in ASTCFLAGS
  and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
  by the build system itself, so that the user can effectively override the
  build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
  be provided *either* in the environment before running 'make', or as variable
  assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
  is no longer necessary, so they are no longer documented, but are still supported
  so as not to break existing build systems that supply them when building Asterisk.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 207573 via svnmerge
Jeff Peeler [Mon, 20 Jul 2009 23:31:36 +0000 (23:31 +0000)]
Blocked revisions 207573 via svnmerge

........
  r207573 | jpeeler | 2009-07-20 18:23:18 -0500 (Mon, 20 Jul 2009) | 10 lines

  Wait for wink before dialing when using E&M wink signaling

  This patch adds a new dahdi_wait function to specifically wait for the wink
  event. If the wink is not eventually received the channel is hung up.

  (closes issue #14434)
  Reported by: araasch
  Patches:
        emwinkmod uploaded by araasch (license 693)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207599 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoOkay, that didn't fix the crash. It didn't really do anything useful.
Mark Michelson [Mon, 20 Jul 2009 23:08:56 +0000 (23:08 +0000)]
Okay, that didn't fix the crash. It didn't really do anything useful.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoInitialize connected line instance when doing a directed pickup.
Mark Michelson [Mon, 20 Jul 2009 22:13:34 +0000 (22:13 +0000)]
Initialize connected line instance when doing a directed pickup.

This helps to prevent a crash which may occur due to our freeing
garbage due to a struct being uninitialized.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoreg->username is parsed only once on sip reload
David Vossel [Mon, 20 Jul 2009 20:45:26 +0000 (20:45 +0000)]
reg->username is parsed only once on sip reload

The registration string can contain an expanded user portion of the
form user@domain. This expanded user portion was stored in
reg->username and parsed each time there is a registration refresh.
Now, the domain portion of the user is parsed and stored separately
in the regdomain field.

(closes issue #14331)
Reported by: Nick_Lewis
Patches:
      chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207423 via svnmerge from
Mark Michelson [Mon, 20 Jul 2009 19:48:12 +0000 (19:48 +0000)]
Merged revisions 207423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines

  Answer video SDP offers properly when videosupport is not enabled.

  Copied from Review board:

  In issue 12434, the reporter describes a situation in which audio and video
  is offered on the call, but because videosupport is disabled in sip.conf,
  Asterisk gives no response at all to the video offer. According to RFC 3264,
  all media offers should have a corresponding answer. For offers we do not
  intend to actually reply to with meaningful values, we should still reply
  with the port for the media stream set to 0.

  In this patch, we take note of what types of media have been offered and
  save the information on the sip_pvt. The SDP in the response will take into
  account whether media was offered. If we are not otherwise going to answer
  a media offer, we will insert an appropriate m= line with the port set to 0.

  It is important to note that this patch is pretty much a bandage being
  applied to a broken bone. The patch *only* helps for situations where video
  is offered but videosupport is disabled and when udptl_pt is disabled but
  T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
  Notable cases are when multiple streams of the same type are offered.
  The 2 media stream limit is still present with this patch, too.

  In trunk and the 1.6.X branches, things will be a bit different since Asterisk
  also supports text in SDPs as well.

  (closes issue #12434)
  Reported by: mnnojd

  Review: https://reviewboard.asterisk.org/r/311
  Review: https://reviewboard.asterisk.org/r/313
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207360 via svnmerge from
Russell Bryant [Mon, 20 Jul 2009 16:36:15 +0000 (16:36 +0000)]
Merged revisions 207360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines

  Only do the chan->fdno check in ast_read() in a developer build.

  I changed this check to only happen in a dev-mode build.  I also added a
  comment explaining what is going on.  I also made it so that detection of
  this situation does not affect ast_read() operation.

  (closes issue #14723)
  Reported by: seadweller
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207361 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged 207316 from
Richard Mudgett [Sat, 18 Jul 2009 04:17:01 +0000 (04:17 +0000)]
Merged 207316 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...

..........
r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) | 20 lines

Fixed incoming calls being matched to MSNs without type-of-number prefix added.

For an incoming ISDN call the dialed.number is incorrectly matched against
the configured MSNs in misdn.conf.  The numbers passed to the dialplan
include the configured prefix for the dialed.number_type, whereas the
check against the configured MSNs (to decide if the call is accepted at
all), is executed without the configured prefix.

e.g., dialed.number = 241168020, TON = national, configured national
prefix is "0".  (This is the TON which is used by ISDN providers in the
Netherlands.)

In chan_misdn.c:cb_events() in case EVENT_SETUP the call to
misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57
lines later the call to read_config() adds the prefix, and the
dialed.number is now 0241168020, which is then used in the dialplan.
misdn_cfg_is_msn_valid() must use the normalized number, too.

JIRA ABE-1912

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFlag field in wrong position.
Tilghman Lesher [Sat, 18 Jul 2009 04:16:44 +0000 (04:16 +0000)]
Flag field in wrong position.
Reported by "Hoggins!" on asterisk-dev list.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRecorded merge of revisions 145293,158010 via svnmerge from
Richard Mudgett [Sat, 18 Jul 2009 01:31:53 +0000 (01:31 +0000)]
Recorded merge of revisions 145293,158010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines

  channels/chan_misdn.c
  channels/misdn/isdn_lib.c
  *  Miscellaneous other fixes from trunk to make merging easier later.

  ........
  r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines

  *  Miscellaneous formatting changes to make v1.4 and trunk
  more merge compatible in the mISDN area.

  channels/chan_misdn.c
  *  Eliminated redundant code in cb_events() EVENT_SETUP

  ........
  r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines

  improved helptext of misdn_set_opt.
  ........
  r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line

  Cleaned up comment

  ........
  r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines

  channels/chan_misdn.c
  *  Made bearer2str() use allowed_bearers_array[]
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Made use Asterisk presentation indicator values if either of the
  mISDN presentation or screen options are negative.
  *  Updated the misdn_set_opt application option descriptions.
  *  Renamed the awkward Caller ID presentation misdn_set_opt
  application option value not_screened to restricted.
  Deprecated the not_screened option value.

  channels/misdn/isdn_lib.c
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Fixed some spelling errors and typos.
  *  Added all defined facility code strings to fac2str().

  channels/misdn/isdn_lib.h
  *  Added doxygen comments to struct misdn_bchannel.

  channels/misdn/isdn_lib_intern.h
  *  Added doxygen comments to struct misdn_stack.

  channels/misdn_config.c
  configs/misdn.conf.sample
  *  Updated the mISDN presentation and screen parameter descriptions.

  doc/misdn.txt (doc/tex/misdn.tex)
  *  Updated the misdn_set_opt application option descriptions.
  *  Fixed some spelling errors and typos.
................
  r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines

  Merged revision 157977 from
  https://origsvn.digium.com/svn/asterisk/team/group/issue8824

  ........
  Fixes JIRA ABE-1726

  The dial extension could be empty if you are using MISDN_KEYPAD
  to control ISDN provider features.
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd flag here, too (as requested by jsmith)
Tilghman Lesher [Fri, 17 Jul 2009 22:29:50 +0000 (22:29 +0000)]
Add flag here, too (as requested by jsmith)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207255 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agofixes an error in r203638 CEL commit
David Vossel [Fri, 17 Jul 2009 22:07:36 +0000 (22:07 +0000)]
fixes an error in r203638 CEL commit

(closes issue #15525)
Reported by: elguero
Patches:
      iax2-double-unlock.patch uploaded by elguero (license 37)
      15525.diff uploaded by dvossel (license 671)
Tested by: dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoDocument the "flag" field in the voicemessages table.
Tilghman Lesher [Fri, 17 Jul 2009 22:04:43 +0000 (22:04 +0000)]
Document the "flag" field in the voicemessages table.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 207155 via svnmerge from
Jeff Peeler [Fri, 17 Jul 2009 19:37:38 +0000 (19:37 +0000)]
Merged revisions 207155 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines

  Fix format specifier to print out an unsigned long long.

  Yep, it's even ifdefed out code. But it made it to the RR list...

  (closes issue #14726)
  Reported by: lmadsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207156 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoUpdate some missing allowed options for overlapdial
Jeff Peeler [Fri, 17 Jul 2009 19:16:35 +0000 (19:16 +0000)]
Update some missing allowed options for overlapdial

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 207092 via svnmerge
Jeff Peeler [Fri, 17 Jul 2009 19:14:02 +0000 (19:14 +0000)]
Blocked revisions 207092 via svnmerge

........
  r207092 | jpeeler | 2009-07-17 14:13:27 -0500 (Fri, 17 Jul 2009) | 11 lines

  Enhance configuration option for overlapdial allowing direction choice

  Previously overlap dialing could only be turned on or off for both incoming and
  outgoing calls. New parameters incoming, outgoing, and both have been added to
  allow further control. There is no change in default behavior with these new
  options and allows in band DTMF to be accepted in one direction if required.

  (closes issue #14471)
  Reported by: eboscani
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207093 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 207033 via svnmerge
David Vossel [Fri, 17 Jul 2009 18:01:04 +0000 (18:01 +0000)]
Blocked revisions 207033 via svnmerge

........
  r207033 | dvossel | 2009-07-17 13:00:38 -0500 (Fri, 17 Jul 2009) | 4 lines

  sip option flags handled incorrectly

  (issue #15376)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agosip option flags handled incorrectly
David Vossel [Fri, 17 Jul 2009 17:51:44 +0000 (17:51 +0000)]
sip option flags handled incorrectly

(closes issue #15376)
Reported by: Takehiko Ooshima
Tested by: dvossel, Takehiko_Ooshima

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix segfault in sig_analog when using callwaiting, respect callwaiting options
Jeff Peeler [Fri, 17 Jul 2009 17:02:44 +0000 (17:02 +0000)]
Fix segfault in sig_analog when using callwaiting, respect callwaiting options

Sig_analog handles allocating the sub channel for callwaiting, so no longer try
to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as
allocated upon success of the alloc_sub callback, which was responsible for the
segfault. Also, the callwaiting and callwaitingcallerid options were being
unconditionally set to true. Now, the options are properly set from
chan_dahdi.conf.

(closes issue #15508)
Reported by: elguero
Tested by: elguero

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206938 via svnmerge from
David Vossel [Fri, 17 Jul 2009 16:13:22 +0000 (16:13 +0000)]
Merged revisions 206938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines

  SIP incorrect From: header information when callpres is prohib

  Some ITSP make use of the "Anonymous" display name to detect a
  requirement to withhold caller id across the PSTN. This does
  not work if the display name is "Unknown".

  (closes issue #14465)
  Reported by: Nick_Lewis
  Patches:
        chan_sip.c-callerpres.patch uploaded by Nick (license 657)
        chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
  Tested by: Nick_Lewis, dvossel
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoTIMEOUT(absolute) returned negative value.
David Vossel [Thu, 16 Jul 2009 21:45:14 +0000 (21:45 +0000)]
TIMEOUT(absolute) returned negative value.

(closes issue #15513)
Reported by: ys

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206872 via svnmerge from
David Vossel [Thu, 16 Jul 2009 21:33:51 +0000 (21:33 +0000)]
Merged revisions 206872 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines

  error in iax.conf related IP-based access control

  (closes issue #15518)
  Reported by: pkempgen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206873 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206867 via svnmerge from
David Vossel [Thu, 16 Jul 2009 21:25:22 +0000 (21:25 +0000)]
Merged revisions 206867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines

  avoid segfault caused by user error

  If the CALLERPRES() dialplan function is set to nothing,
  a segfault occurs.  This is user error to begin with, but
  I'd rather see a cli warning message than have Asterisk
  crash on me.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206807 via svnmerge from
Tilghman Lesher [Thu, 16 Jul 2009 16:51:05 +0000 (16:51 +0000)]
Merged revisions 206807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines

  Fix a memory leak.
  (closes issue #15517)
   Reported by: adomjan
   Patches:
         func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoSession timer were not activated if Supported header field in INVITE had both "timer...
David Vossel [Wed, 15 Jul 2009 22:04:13 +0000 (22:04 +0000)]
Session timer were not activated if Supported header field in INVITE had both "timer" and other options.

(closes issue #15403)
Reported by: makoto
Patches:
      sip-session-timer.patch uploaded by makoto (license 38)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoThe dialing flag was mistakingly removed from sig_pri.
Jeff Peeler [Wed, 15 Jul 2009 22:02:55 +0000 (22:02 +0000)]
The dialing flag was mistakingly removed from sig_pri.

This readds the proper setting of the flag and is really a continuation of
r205731. The flag was being set properly in sig_analog, but use of the
newly added set_dialing callback allowed for some simplification in
chan_dahdi.

(closes issue #15486)
Reported by: rmudgett

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206706 via svnmerge from
Richard Mudgett [Wed, 15 Jul 2009 21:14:41 +0000 (21:14 +0000)]
Merged revisions 206706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines

  Merged revision 206700 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...

  ..........
    Fixed chan_misdn crash because mISDNuser library is not thread safe.

    With Asterisk the mISDNuser library is driven by two threads concurrently:
    1. channels/misdn/isdn_lib.c::manager_event_handler()
    2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()

    Calls into the library are done concurrently and recursively from
    isdn_lib.c.

    Both threads can fiddle with the master/child layer3_proc_t lists.  One
    thread may traverse the list when the other interrupts it and then removes
    the list element which the first thread was currently handling.  This is
    exactly what caused the crash.  About 60 calls were needed to a Gigaset
    CX475 before it occurred once.

    This patch adds locking when calling into the mISDNuser library.
    This also fixes some cb_log calls with wrong port parameter.

    JIRA ABE-1913
        Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
  ..........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agocallerid(num) is wrong when username is missing
David Vossel [Wed, 15 Jul 2009 20:20:01 +0000 (20:20 +0000)]
callerid(num) is wrong when username is missing

A domain only sip uri <sip:123.123.123.123> would return
123.123.123.123 as callid num.  Now, if the username is
missing from a uri, the callerid num field is left empty.

(closes issue #15476)
Reported by: viraptor

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206702 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206635 via svnmerge from
Sean Bright [Wed, 15 Jul 2009 16:00:24 +0000 (16:00 +0000)]
Merged revisions 206635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line

  Only print debug info in codec_dahdi if we are asking for it.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agofix a typo in sample config file for option change
Jeff Peeler [Tue, 14 Jul 2009 20:38:56 +0000 (20:38 +0000)]
fix a typo in sample config file for option change

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoDocument all meetme realtime fields, and in the process, make some field lengths...
Tilghman Lesher [Tue, 14 Jul 2009 20:14:45 +0000 (20:14 +0000)]
Document all meetme realtime fields, and in the process, make some field lengths more consistent.
(closes issue #15493)
 Reported by: lasko
 Patches:
       meetme.diff uploaded by lasko (license 833)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRestore some missing functionality to sig_analog.
Jeff Peeler [Tue, 14 Jul 2009 20:01:10 +0000 (20:01 +0000)]
Restore some missing functionality to sig_analog.

The main purpose of this commit is to restore missing functionality present in
the ss_thread before all the sig related work was done. Two of the biggest
missing things were distinctive ring detection and cid handling for V23.
fxsoffhookstate and associated mwi variables have been moved inside sig_analog
as they were not being set properly as well.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoI AM A TERRIBLE PERSON
Mark Michelson [Tue, 14 Jul 2009 17:03:58 +0000 (17:03 +0000)]
I AM A TERRIBLE PERSON

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206487 via svnmerge from
Richard Mudgett [Tue, 14 Jul 2009 17:01:48 +0000 (17:01 +0000)]
Merged revisions 206487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines

  Fixes several call transfer issues with chan_misdn.

  *  issue #14355 - Crash if attempt to transfer a call to an application.
  Masquerade the other pair of the four asterisk channels involved in the
  two calls.  The held call already must be a bridged call (not an
  applicaton) or it would have been rejected.

  *  issue #14692 - Held calls are not automatically cleared after transfer.
  Allow the core to initate disconnect of held calls to the ISDN port.  This
  also fixes a similar case where the party on hold hangs up before being
  transferred or taken off hold.

  *  JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
  Do not simply block passing the hangup event on held calls to asterisk
  core.

  *  Fixed to allow held calls to be transferred to ringing calls.
  Previously, held calls could only be transferred to connected calls.
  *  Eliminated unused call states to simplify hangup code.
  *  Eliminated most uses of "holded" because it is not a word.

  (closes issue #14355)
  (closes issue #14692)
  Reported by: sodom
  Patches:
        misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoReset the sentringing indication when redirects occur.
Mark Michelson [Tue, 14 Jul 2009 16:09:38 +0000 (16:09 +0000)]
Reset the sentringing indication when redirects occur.

If a redirecting control frame is processed or a call forward occurs,
we need to reset the sentringing flag so that we can send another ringing
indication to the phone that may contain a connected line update.

AST-164

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206455 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206385 via svnmerge from
Russell Bryant [Tue, 14 Jul 2009 14:51:44 +0000 (14:51 +0000)]
Merged revisions 206385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines

  Merged revisions 206384 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.2

  ........
    r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines

    Ensure apathetic replies are sent out on the proper socket.

    chan_iax2 supports multiple address bindings.  The send_apathetic_reply()
    function did not attempt to send its response on the same socket that the
    incoming message came in on.
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 206284 via svnmerge from
Richard Mudgett [Tue, 14 Jul 2009 00:48:59 +0000 (00:48 +0000)]
Merged revisions 206284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines

  Fix some memory leaks in chan_misdn.

  JIRA ABE-1911
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agodns lookup of peername rather than peer's host in transmit_register()
David Vossel [Mon, 13 Jul 2009 23:26:51 +0000 (23:26 +0000)]
dns lookup of peername rather than peer's host in transmit_register()

(closes issue #15052)
Reported by: fsantulli
Patches:
      chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818)
Tested by: fsantulli

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206280 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake sure that since we are passing -c to asterisk that we have a console.
Sean Bright [Mon, 13 Jul 2009 18:46:47 +0000 (18:46 +0000)]
Make sure that since we are passing -c to asterisk that we have a console.

Without this line, Asterisk will busy-loop trying to read and write to
/dev/null (woops... my bad).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRemove reference to non-existent help file
Tilghman Lesher [Mon, 13 Jul 2009 16:23:07 +0000 (16:23 +0000)]
Remove reference to non-existent help file
(closes issue #15427)
 Reported by: brushtyler
 Patches:
       app_voicemail.c.diff uploaded by brushtyler (license 821)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206185 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 206126 via svnmerge
Russell Bryant [Mon, 13 Jul 2009 15:12:31 +0000 (15:12 +0000)]
Blocked revisions 206126 via svnmerge

........
  r206126 | russell | 2009-07-13 10:12:08 -0500 (Mon, 13 Jul 2009) | 7 lines

  Print CID match in "show dialplan".

  (closes issue #14702)
  Reported by: klaus3000
  Patches:
        patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000 (license 65)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBump up cleancount so that existing checkouts will update themselves properly for...
Kevin P. Fleming [Mon, 13 Jul 2009 14:06:37 +0000 (14:06 +0000)]
Bump up cleancount so that existing checkouts will update themselves properly for the 'Addons' -> 'ADDONS' change.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206094 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMake the menuselect category for Add-Ons consistent with the other directories (upper...
Kevin P. Fleming [Mon, 13 Jul 2009 13:29:23 +0000 (13:29 +0000)]
Make the menuselect category for Add-Ons consistent with the other directories (uppercase).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agonote the security events API in CHANGES
Russell Bryant [Sat, 11 Jul 2009 19:30:19 +0000 (19:30 +0000)]
note the security events API in CHANGES

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206049 65c4cc65-6c06-0410-ace0-fbb531ad65f3