asterisk/asterisk.git
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

10 years agoAdd an API for reporting security events, and a security event logging module.
Russell Bryant [Sat, 11 Jul 2009 19:15:03 +0000 (19:15 +0000)]
Add an API for reporting security events, and a security event logging module.

This commit introduces the security events API.  This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication.  These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.

Inside of Asterisk, the events go through the ast_event API.  This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.

One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level.  Using logger.conf, these log entries may be
sent to a file, or to syslog.

One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip.  That will be more complicated and will
be done as its own project as the next phase of security events work.

For more information on the security events framework, see the documentation
generated from doc/tex/.  "make asterisk.pdf"

Review: https://reviewboard.asterisk.org/r/273/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoSIP register not using peer's outbound proxy
David Vossel [Fri, 10 Jul 2009 21:42:10 +0000 (21:42 +0000)]
SIP register not using peer's outbound proxy

If callbackextension is defined for a peer it successfully causes
a registration to occur, but the registration ignores the
outboundproxy settings for the peer.  This patch allows the
peer to be passed to obproxy_get() in transmit_register().

(closes issue #14344)
Reported by: Nick_Lewis
Patches:
      callbackextension_peer_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel

Review: https://reviewboard.asterisk.org/r/294/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoUpdate comments about the level of T.38 support in Asterisk.
Kevin P. Fleming [Fri, 10 Jul 2009 18:44:09 +0000 (18:44 +0000)]
Update comments about the level of T.38 support in Asterisk.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205877 via svnmerge from
Mark Michelson [Fri, 10 Jul 2009 17:39:57 +0000 (17:39 +0000)]
Merged revisions 205877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines

  Merged revisions 205776 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/trunk

  ................
    r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines

    Merged revisions 205775 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines

      Ensure that outbound NOTIFY requests are properly routed through stateful proxies.

      With this change, we make note of Record-Route headers present in any SUBSCRIBE
      request that we receive so that our outbound NOTIFY requests will have the proper
      Route headers in them.

      (closes issue #14725)
      Reported by: ibc
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205878 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205804 via svnmerge from
David Vossel [Fri, 10 Jul 2009 16:42:04 +0000 (16:42 +0000)]
Merged revisions 205804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines

  SIP registration auth loop caused by stale nonce

  If an endpoint sends two registration requests in a very short
  period of time with the same nonce, both receive 401 responses
  from Asterisk, each with a different nonce (the second 401
  containing the current nonce and the first one being stale).
  If the endpoint responds to the first 401, it does not match
  the current nonce so Asterisk sends a third 401 with a newly
  generated nonce (which updates the current nonce)... Now if
  the endpoint responds to the second 401, it does not match the
  current nonce either and Asterisk sends a fourth 401 with a
  newly generated nonce... This loop goes on and on.

  There appears to be a simple fix for this.  If the nonce from
  the request does not match our nonce, but is a good response
  to a previous nonce, instead of sending a 401 with a newly
  generated nonce, use the current one instead.  This breaks
  the loop as the nonce is not updated until a response is
  received. Additional logic has been added to make sure no
  nonce can be responded to twice though.

  (closes issue #15102)
  Reported by: Jamuel
  Patches:
        patch-bug_0015102 uploaded by Jamuel (license 809)
        nonce_sip.diff uploaded by dvossel (license 671)
  Tested by: Jamuel

  Review: https://reviewboard.asterisk.org/r/289/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205840 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoEliminate extraneous LOG_DEBUG messages generated by app_fax.
Kevin P. Fleming [Fri, 10 Jul 2009 16:00:44 +0000 (16:00 +0000)]
Eliminate extraneous LOG_DEBUG messages generated by app_fax.

The transmit_audio() and transmit_t38() functions in app_fax have processing
loops that are supposed to wait for frames to arrive on the channel and then
handle them, but they also have short timeouts so that the loops can have
watchdog timers and do other required processing. This commit changes the loops
to not actually call ast_read() and attempt to process the returned frame
unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages
and slightly improving performance.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205780 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205775 via svnmerge from
Mark Michelson [Fri, 10 Jul 2009 15:56:45 +0000 (15:56 +0000)]
Merged revisions 205775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines

  Ensure that outbound NOTIFY requests are properly routed through stateful proxies.

  With this change, we make note of Record-Route headers present in any SUBSCRIBE
  request that we receive so that our outbound NOTIFY requests will have the proper
  Route headers in them.

  (closes issue #14725)
  Reported by: ibc
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix some remaining T.38 negotiation problems in app_fax.
Kevin P. Fleming [Fri, 10 Jul 2009 15:28:11 +0000 (15:28 +0000)]
Fix some remaining T.38 negotiation problems in app_fax.

Revision 205696 did not quite fix all the issues with the T.38 negotiation
changes and app_fax; this patch corrects them, along with a couple of other
minor issues.

(closes issue #15480)
Reported by: dimas
Patches:
      test2-15480.patch uploaded by dimas (license 88)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix mbl_fixup() in chan_mobile to update newchan->tech_pvt instead of oldchan.
Matthew Nicholson [Thu, 9 Jul 2009 21:32:31 +0000 (21:32 +0000)]
Fix mbl_fixup() in chan_mobile to update newchan->tech_pvt instead of oldchan.

(closes issue #15299)
Reported by: nikkk

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205700 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoRepair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Kevin P. Fleming [Thu, 9 Jul 2009 21:20:23 +0000 (21:20 +0000)]
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.

Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).

This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.

(closes issue #14849)
Reported by: afosorio

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoConvert func_odbc to use ast_dummy_alloc_channel()
Matthew Nicholson [Thu, 9 Jul 2009 20:04:43 +0000 (20:04 +0000)]
Convert func_odbc to use ast_dummy_alloc_channel()

Review: https://reviewboard.asterisk.org/r/290/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205666 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205599 via svnmerge from
David Vossel [Thu, 9 Jul 2009 16:19:09 +0000 (16:19 +0000)]
Merged revisions 205599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines

  Changing ast_samp2tv to not use floating point.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agomake this compile again under devmode
Michiel van Baak [Thu, 9 Jul 2009 14:10:01 +0000 (14:10 +0000)]
make this compile again under devmode

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205562 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agopthread_self returns a pthread_t which is not an unsigned int on all
Michiel van Baak [Thu, 9 Jul 2009 08:31:24 +0000 (08:31 +0000)]
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.

Tested on OpenBSD and Linux both 32 and 64 bit

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205471 via svnmerge from
David Vossel [Wed, 8 Jul 2009 23:19:09 +0000 (23:19 +0000)]
Merged revisions 205471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines

  Fixes 8khz assumptions

  Many calculations assume 8khz is the codec rate. This
  is not always the case.  This patch only addresses chan_iax.c
  and res_rtp_asterisk.c, but I am sure there are other areas
  that make this assumption as well.

  Review: https://reviewboard.asterisk.org/r/306/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix a CEL related regression with hints updating by subscribing to AST_DEVICE_STATE...
Matthew Nicholson [Wed, 8 Jul 2009 23:07:09 +0000 (23:07 +0000)]
Fix a CEL related regression with hints updating by subscribing to AST_DEVICE_STATE instead of AST_DEVICE_STATE_CHANGED.

(closes issue #15440)
Reported by: lmsteffan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205409 via svnmerge from
David Vossel [Wed, 8 Jul 2009 22:15:06 +0000 (22:15 +0000)]
Merged revisions 205409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines

  moving ast_devstate_to_extenstate to pbx.c from devicestate.c

  ast_devstate_to_extenstate belongs in pbx.c.  This change
  fixes a compile time error with chan_vpb as well.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205412 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agomissing comma in devstatestring array
David Vossel [Wed, 8 Jul 2009 22:02:54 +0000 (22:02 +0000)]
missing comma in devstatestring array

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205410 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205349 via svnmerge from
Mark Michelson [Wed, 8 Jul 2009 19:26:55 +0000 (19:26 +0000)]
Merged revisions 205349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines

  Prevent phantom calls to queue members.

  If a caller were to hang up while a periodic announcement or position
  were being said, the return value for those functions would incorrectly
  indicate that the caller was still in the queue. With these changes,
  the problem does not occur.

  (closes issue #14631)
  Reported by: latinsud
  Patches:
        queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
     (with small modification from me)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205288 via svnmerge from
Jason Parker [Wed, 8 Jul 2009 18:19:46 +0000 (18:19 +0000)]
Merged revisions 205288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line

  Update config.guess and config.sub from the savannah.gnu.org git repo.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205291 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFixes Park() argument handling
David Brooks [Wed, 8 Jul 2009 17:26:26 +0000 (17:26 +0000)]
Fixes Park() argument handling

Park() was not respecting the arguments passed to it. Any extension/context/priority
given to it was being ignored. This patch remedies this.

(closes issue #15380)
Reported by: DLNoah

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoOops, fixing build
Tilghman Lesher [Wed, 8 Jul 2009 16:59:32 +0000 (16:59 +0000)]
Oops, fixing build

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205215 via svnmerge from
David Vossel [Wed, 8 Jul 2009 16:54:24 +0000 (16:54 +0000)]
Merged revisions 205215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines

  ast_samp2tv needs floating point for 16khz audio

  In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
  The .5 is currently stripped off because we don't calculate
  using floating points.  This causes madness with 16khz audio.

  (issue ABE-1899)

  Review: https://reviewboard.asterisk.org/r/305/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205216 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix a few compilation problems found when building Asterisk against uClibc.
Sean Bright [Wed, 8 Jul 2009 16:43:12 +0000 (16:43 +0000)]
Fix a few compilation problems found when building Asterisk against uClibc.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205214 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 205188 via svnmerge from
Tilghman Lesher [Wed, 8 Jul 2009 16:27:50 +0000 (16:27 +0000)]
Merged revisions 205188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines

  Add redirection warnings for the invalid language codes previously removed.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoUse tabs instead of spaces for indentation.
Russell Bryant [Wed, 8 Jul 2009 15:56:28 +0000 (15:56 +0000)]
Use tabs instead of spaces for indentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoBlocked revisions 205149 via svnmerge
Russell Bryant [Wed, 8 Jul 2009 15:54:42 +0000 (15:54 +0000)]
Blocked revisions 205149 via svnmerge

........
  r205149 | russell | 2009-07-08 10:54:21 -0500 (Wed, 08 Jul 2009) | 8 lines

  Make OpenSSL usage thread-safe.

  OpenSSL is not thread-safe by default.  However, making it thread safe is
  very easy.  We just have to provide a couple of callbacks.  One callback
  returns a thread ID.  The other handles locking.  For more information,
  start with the "Is OpenSSL thread-safe?" question on the FAQ page of
  openssl.org.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205150 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMove OpenSSL initialization to a single place, make library usage thread-safe.
Russell Bryant [Wed, 8 Jul 2009 15:17:19 +0000 (15:17 +0000)]
Move OpenSSL initialization to a single place, make library usage thread-safe.

While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.

1) We had initialization of the library done in multiple modules.  This has now
   been moved to a core function that gets executed during Asterisk startup.
   We already link OpenSSL into the core for TCP/TLS functionality, so this
   was the most logical place to do it.

2) OpenSSL is not thread-safe by default.  However, making it thread safe is
   very easy.  We just have to provide a couple of callbacks.  One callback
   returns a thread ID.  The other handles locking.  For more information,
   start with the "Is OpenSSL thread-safe?" question on the FAQ page of
   openssl.org.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFreeBSD now has autoconf 2.62 in the ports, 2.61 has disappeared.
Luigi Rizzo [Wed, 8 Jul 2009 14:45:15 +0000 (14:45 +0000)]
FreeBSD now has autoconf 2.62 in the ports, 2.61 has disappeared.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoPermit setting custom headers from the peer definition.
Tilghman Lesher [Tue, 7 Jul 2009 21:10:14 +0000 (21:10 +0000)]
Permit setting custom headers from the peer definition.
(closes issue #14059)
 Reported by: fnordian

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoFix a deadlock in sig_analog
Matthew Nicholson [Tue, 7 Jul 2009 18:24:13 +0000 (18:24 +0000)]
Fix a deadlock in sig_analog

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205047 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd CEL transfer events to analog (chan_dahdi) transfers.
Matthew Nicholson [Mon, 6 Jul 2009 23:24:57 +0000 (23:24 +0000)]
Add CEL transfer events to analog (chan_dahdi) transfers.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 981 via svnmerge from
Tilghman Lesher [Mon, 6 Jul 2009 21:37:39 +0000 (21:37 +0000)]
Merged revisions 981 via svnmerge from
https://origsvn.digium.com/svn/asterisk-addons/branches/1.4

........
  r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul 2009) | 7 lines

  Don't reset reconnect time, unless a reconnect really occurred.
  (closes issue #15375)
   Reported by: kowalma
   Patches:
         20090628__issue15375.diff.txt uploaded by tilghman (license 14)
   Tested by: kowalma, jacco
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204986 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoImprove handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capab...
Kevin P. Fleming [Mon, 6 Jul 2009 13:38:29 +0000 (13:38 +0000)]
Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels.

This change allows applications that request T.38 negotiation on a channel that
does not support it to get the proper indication that it is not supported, rather
than thinking that negotiation was started when it was not.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoAdd a configure check for Reverse Charging Indication support in LibPRI.
Sean Bright [Fri, 3 Jul 2009 15:44:01 +0000 (15:44 +0000)]
Add a configure check for Reverse Charging Indication support in LibPRI.

Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoWrap rtp_engine.h header comments to 80 characters.
Sean Bright [Fri, 3 Jul 2009 02:02:50 +0000 (02:02 +0000)]
Wrap rtp_engine.h header comments to 80 characters.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMerged revisions 204834 via svnmerge from
Richard Mudgett [Thu, 2 Jul 2009 22:01:28 +0000 (22:01 +0000)]
Merged revisions 204834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines

  Removed confusing warning message "Got Busy in Connected State"

  If an incoming mISDN call is answered with the Answer application and a
  subsequent Dial gets a busy endpoint then it is valid for that already
  connected channel to get the busy indication.  Asterisk will play the busy
  tones until the dialplan plays something else or hangs up the call.

  (closes issue #11974)
  Reported by: fvdb
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204835 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoMoved trigger for BRIDGE_END CEL event so that it is more accurate.
Matthew Nicholson [Thu, 2 Jul 2009 20:37:16 +0000 (20:37 +0000)]
Moved trigger for BRIDGE_END CEL event so that it is more accurate.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

10 years agoSupport setting and receiving Reverse Charging Indication over ISDN PRI.
Sean Bright [Thu, 2 Jul 2009 17:46:14 +0000 (17:46 +0000)]
Support setting and receiving Reverse Charging Indication over ISDN PRI.

This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI.  This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:

...,Dial(DAHDI/g1/C4445556666)

And to read it off an inbound channel:

exten => s,1,Set(RCI=${CHANNEL(reversecharge)})

Thanks again to rmudgett for the thorough review.

(closes issue #13760)
Reported by: mrgabu

Review: https://reviewboard.asterisk.org/r/303/

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