Suggested on the -dev list, and implemented in an alternate way by me.


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........
  r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines
  
  make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD
........


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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines

Make apps send PROGRESS control frame for early media and fix too early media issue in SIP

The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to 
play silence and ignore the later 180 ringing message. A bad user experience.

The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
  any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
  before the outbound channel actually indicates any sort of call progress.
  In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
  of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
  phone experience - only for the better.

We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.

This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems). 
That's no proof that this is an excellent patch, but, well, it's tested :-)


........


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  r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009) | 2 lines
  
  make chan_sip compile under devmode again
........


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  r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009) | 4 lines
  
  Make sure 'start' is always initialized.
  
  This is the same as rev 216222 in trunk but 1.4 is affected as well
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There was a problem in the function responsible for doing peer matching by
IP address and port number such that during the second pass for checking for
a peer configured with insecure=port, it would end up treating every peer as
if it had been configured that way.  These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking properly.

This problem was introduced when SIP peers were converted to astobj2.  Many
thanks to dvossel for noticing this while working on another peer matching
issue.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up 
one of these is often a good way to get into Asterisk development and getting a lot of developers in 
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.

If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.

Thank you for your help.

</end of long commit message>


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  r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines
  
  Merged revisions 216262 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines
    
    Add a plain text version of the IAX2 security document.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216264 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Makes asterisk compile with --enable-dev-mode


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216186 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.  

(closes issue #9096)
Reported by: fleed
Patches:
      9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs


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  r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines
  
  Merged revisions 216080 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines
    
    Add a note about IAX2 to UPGRADE.txt.
  ........
................


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  r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines
  
  Merged revisions 216005 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
    
    Add IAX2 security document related to AST-2009-006.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216009 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.

(closes issue #15771)
Reported by: jtodd
Patches:
      language-criteria.txt uploaded by jtodd



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........
  r216000 | dvossel | 2009-09-03 13:32:32 -0500 (Thu, 03 Sep 2009) | 7 lines
  
  Merge code associated with AST-2009-006
  
  (closes issue #12912)
  Reported by: rathaus
  Tested by: tilghman, russell, dvossel, dbrooks
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216001 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14573)
Reported by: pj
Patches: 
      sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
	modified by oej
Tested by: pj



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

the configured parkinglot in their response.

Prodded by snuff-work on #asterisk-dev IRC channel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15764)
 Reported by: elguero
 Change-type: bugfix


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This did not function in the way that was intended, causing more compatibility
issues than it solved.  It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)


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........
  r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines
  
  Re-send non-100 provisional responses to prevent cancellation
  
  From section 13.3.1.1 of RFC 3261:
  
     If the UAS desires an extended period of time to answer the INVITE,
     it will need to ask for an "extension" in order to prevent proxies
     from canceling the transaction. A proxy has the option of canceling
     a transaction when there is a gap of 3 minutes between responses in a
     transaction. To prevent cancellation, the UAS MUST send a non-100
     provisional response at every minute, to handle the possibility of
     lost provisional responses.
  
  (closes issue #11157)
  Reported by: rjain
  Tested by: twilson
  
  Review: https://reviewboard.asterisk.org/r/315/
........


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There are several instances where a port is parsed
from a uri or some other source and converted to
an int value using atoi(), if for some reason the
port string is empty, then a standard port is used.
This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215681 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215665 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

made analog_set_linear_mode return back the mode that was being overwritten 
so it could be restored later. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also, a Makefile fix for Darwin (OS X).
(closes issue #14542)
 Reported by: jtodd
 Patches: 
       20090901__issue14542.diff.txt uploaded by tilghman (license 14)
 Tested by: jtodd, tilghman
 Change-type: bugfix


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Now, the scheme passed to parse_uri can either be a
single scheme, or a list of schemes ',' delimited.
This gets rid of the whole problem of having to create
two buffers and calling parse_uri twice to check for
separate schemes.

Review: https://reviewboard.asterisk.org/r/343/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny lines as well.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

keep-alive events are used by Sipura/Linksys for NAT keepalive.
There currently don't appear to be any problems with NAT, but
everytime a keep-alive event is received, Asterisk responds with a
"489 Bad event".  This error may indicate to a user that NAT
problems exist just because this even is not supported.  Now,
rather than respond with an error, the packet is consumed and
a "200 ok" is sent just to indicate we received the packet.

(issue #15084)
Patches:
      chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer
into the newly created channel.

(closes issue #15538)
Reported by: gracedman
Patches:
      2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7)
	  With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/345/



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

........
  r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
  
  Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
  
  In general channel names are in the form Foo/Bar-Z, but the channel name
  could have multiple hyphens and look like Foo/B-a-r-Z.  Use strrchr to
  truncate the channel name at the last hyphen.
  
  (closes issue #15810)
  Reported by: dhubbard
  Patches:
        dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13140)
 Reported by: cpina
 Patches: 
       20090807__issue13140.diff.txt uploaded by tilghman (license 14)
 Tested by: lmadsen
 Change-type: feature


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(AST-228)


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format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames returned by
read().  However, it lied.  This means that other parts of the code that
attempted to make use of the offset buffer would end up corrupting the fields
in the ast_filestream structure.  This resulted in quite a few crashes due to
unexpected values for fields in ast_filestream.

This patch closes out quite a few bugs.  However, some of these bugs have been
open for a while and have been an area where more than one bug has been
discussed.  So with that said, anyone that is following one of the issues
closed here, if you still have a problem, please open a new bug report for the
specific problem you are still having.  If you do, please ensure that the bug
report is based on the newest version of Asterisk, and that this patch is
applied if format_mp3 is in use.  Thanks!

(closes issue #15109)
Reported by: jvandal
Tested by: aragon, russell, zerohalo, marhbere, rgj

(closes issue #14958)
Reported by: aragon

(closes issue #15123)
Reported by: axisinternet

(closes issue #15041)
Reported by: maxnuv

(closes issue #15396)
Reported by: aragon

(closes issue #15195)
Reported by: amorsen
Tested by: amorsen

(closes issue #15781)
Reported by: jensvb

(closes issue #15735)
Reported by: thom4fun

(closes issue #15460)
Reported by: marhbere


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decoded.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215110 65c4cc65-6c06-0410-ace0-fbb531ad65f3