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.


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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



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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.
  ........
................


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

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  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

Bump up cleancount so that existing checkouts will update themselves properly for the 'Addons' -> 'ADDONS' change.



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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

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/



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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
    ........
  ................
................


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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

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

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

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

(closes issue #15299)
Reported by: nikkk


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

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

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


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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.
........


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pthread implementations. Casting it to an unsigned int fixes compiler warnings.

Tested on OpenBSD and Linux both 32 and 64 bit


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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

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


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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.
........


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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)
........


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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

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


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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/
........


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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.
........


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