........
  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

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

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

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

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

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

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

Reported by "Hoggins!" on asterisk-dev list.


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

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

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

(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

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

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

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

........
  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

........
  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

(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

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

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

(closes issue #15513)
Reported by: ys



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

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

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

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

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

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

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

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

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

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

(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

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

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

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

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

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

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

(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

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

(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

........
  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