A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.



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(closes issue #15851)
 Reported by: tzafrir


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* Don't write asterisk.conf from scratch. Fix the existing one.
* Pass extra 'make' command-line arguments to 'install' and 'samples'.
* Fix some extra typos.

closes issue #15019 .


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parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.

(closes issue #15839)
Reported by: ebroad
Patches:
      blank_cidv2.patch uploaded by ebroad (license 878)
      parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel



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Please follow the structure of the source code, thanks. Chan_sip is messy enough as it is :-)



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- Remove unused string in sip_registry -- "random"
- Someone added a function in the middle of all forward declarations... Weird. Moved it out of that
  section.


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- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt
- Rename "text" to "codecs" - beacuse it's what it is
- Add documentation for future developers so that we make sure that we define new sdp media types
  for SRTP-variants


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Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher.


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Patch by oej

closes issue #15621
Reported by: fnordian
Tested by: atis


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This patch removes the contact header matching logic and
adds logic to match all tcp/tls connections by ip only.
Thanks to oej for finding the issue and suggesting solutions.

Review: https://reviewboard.asterisk.org/r/354/


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


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

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

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

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

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


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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.


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


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Makes asterisk compile with --enable-dev-mode


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