git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines
  
  Use unsigned ints for portinuri flags.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
  
  Fix SRV lookup and Request-URI generation in chan_sip.
  
  This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct.  That field is used during RURI generation to determine if the port should be included in the RURI.  It is also used in some places to determine if an SRV lookup should occur.
  
  (closes issue #14418)
  Reported by: klaus3000
  Tested by: klaus3000, mnicholson
  
  Review: https://reviewboard.asterisk.org/r/369/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines
  
  check bounds - prevents for buffer overflow
........
  r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines
  
  added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf
  
  (closes issue #15471)
  Reported by: dkerr
  Patches:
        csv_quote_14.txt uploaded by mnick (license )
  Tested by: mnick
........
  r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines
  
  changed the prototype definition of csv_quote
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks, Josh.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
  
  Change the SSRC by default when our media stream changes
  
  Be default, change SSRC when doing an audio stream changes Asterisk doesn't
  honor marker bit when reinvited to already-bridged RTP streams,resulting in
  far-end stack discarding packets with "old" timestamps that areactually part of
  a new stream.  This patch sends AST_CONTROL_SRCUPDATE whenever there is a
  reinvite, unless the 'constantssrc' is set to true in sip.conf.
  
  The original issue reported to Digium support detailed the following situation:
  ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
  fromITSP, Asterisk dials the app server which sends a re-invite back
  toAsterisk--not to negotiate to send media directly to the ITSP, but to
  indicatethat it's changing the stream it's sending to Asterisk.  The app
  servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
  bit on the new stream.  Asterisk passes through the teimstamp of the new stream,
  butdoes not reset the SSRC, sequence numbers, or set the marker bit.
  
  When the timestamp on the new stream is older than the timestamp on the
  originalstream, the ITSP (which doesn't know there has been any change) discards
  the newframes because it thinks they are too old.  This patch addresses this by
  changing the SSRC on a stream update unless constantssrc=true is set in
  sip.conf.
  
  Review: https://reviewboard.asterisk.org/r/374/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
  
  Avoid a potential NULL dereference.
  (closes issue #15865)
   Reported by: kobaz
   Patches: 
         20090915__issue15865.diff.txt uploaded by tilghman (license 14)
   Tested by: kobaz
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14740)
Reported by: pj
Patches:
      issue14740_09022009.diff uploaded by seanbright (license 71)
Tested by: seanbright, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
      __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14859)
 Reported by: atis
 Patches: 
       20090821__issue14859.diff.txt uploaded by tilghman (license 14)
       20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14)
 Tested by: atis, tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221044 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_cel_report_event would geet a reference to the
bridged channel. However, certain return paths, such
as if CEL was not enabled, would result in a reference
leak. All return paths now properly unref the channel.

(closes issue #15991)
Reported by: mmichelson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r220907 | mnicholson | 2009-09-29 15:14:29 -0500 (Tue, 29 Sep 2009) | 10 lines
  
  Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
  
  
  (closes issue #15965)
  Reported by: atis
  Patches:
        chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96)
  Tested by: atis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
  
  Reduce CPU usage related to building a peer merely for devicestates.
  This fixes a 100% CPU problem in the SIP driver, found by profiling
  the driver while the problem was occurring.
  (closes issue #14309)
   Reported by: pkempgen
   Patches: 
         20090924__issue14309.diff.txt uploaded by tilghman (license 14)
   Tested by: pkempgen, vrban
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220906 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix options 'm' and 's'. They were swapped in the code.  Also document the fact that app_confbridge does not automatically answer the channel.

(closes issue #15964)
Reported by: shrift


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!

(closes issue #14949)
Reported by: noahisaac
Patches:
      vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), 
      modified by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines
  
  When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler
  so we override any default optimization levels for a particular install.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.

(closes issue #15943)
Reported by: tpsast



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

*  Removed unneeded and uninitialized service_lock.
*  Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART.
*  Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Pointed out to me on IRC by snuff-home. Thanks


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Including version.h here causes this file to get recompiled after
every commit or update, which is not needed.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based
channel lookups, but this will not work properly when the channel's full
name was not supplied; for name-prefix searches, there is no value in
doing a hash-based lookup, and in fact doing so could result in many
channels being skipped.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).

(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo

Review: https://reviewboard.asterisk.org/r/88/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15939)
Reported by: dvossel

Review: https://reviewboard.asterisk.org/r/375/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.

(closes issue #13165)
Reported by: tim_ringenbach
Patches:
      app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
      modified by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220295 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
  
  Implicitly sending a progress signal breaks some applications.
  Call Progress() in your dialplan if you explicitly want progress to be sent.
  (Reverts change 216430, closes issue #15957)
  Reported by: Pavel Troller on the Asterisk-Dev mailing list
  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line
  
  Resolve parallel build warnings.
  
  Reported by Klaus Darilion on the asterisk-dev mailing list.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines
  
  Remove the remaining bashisms in the Makefile/mkpkgconfig
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220100 65c4cc65-6c06-0410-ace0-fbb531ad65f3