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

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  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

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  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/


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

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

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

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

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

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  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


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

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

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  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

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  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

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

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

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

........
  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

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

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

(closes issue #14059)
 Reported by: fnordian


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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

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

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


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

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

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

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
  
  Improved mapping of extension states from combined device states.
  
  This fixes a few issues with incorrect extension states and adds
  a cli command, core show device2extenstate, to display all possible
  state mappings.
  
  (closes issue #15413)
  Reported by: legart
  Patches:
        exten_helper.diff uploaded by dvossel (license 671)
  Tested by: dvossel, legart, amilcar
  
  Review: https://reviewboard.asterisk.org/r/301/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- cfgbasic.html has been replaced by index.html in the GUI for some time now



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204654 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
  
  More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
  (closes issue #15022)
   Reported by: greenfieldtech
   Patches: 
         20090519__issue15022.diff.txt uploaded by tilghman (license 14)
........


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