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20 authors
  1. Feb 08, 2012
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  15. Jan 31, 2012
  17. Jan 30, 2012
    • Terry Wilson's avatar
      Re-link peers by IP when dnsmgr changes the IP · de57235a
      Terry Wilson authored 
      Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a couple of
issues with this. First, the ast_sockaddr is usually the address of an
ast_sockaddr inside a refcounted struct and we never bump the refcount of those
structs when using dnsmgr. This makes it possible that a refresh could happen
after the destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr cannot be
aware of an address changing without polling for it in the code. If an action
needs to be taken on address update (like re-linking a SIP peer in the
peers_by_ip table), then polling for this change negates many of the benefits
of having dnsmgr in the first place.

This patch adds a function to the dnsmgr API that calls an update callback
instead of blindly updating the address itself. It also moves calls to
ast_dnsmgr_release outside of the destructor functions and into cleanup
functions that are called when we no longer need the objects and increments the
refcount of the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the proper default SIP
port (non-tls vs tls) is also added and used.

This patch also incorporates changes from a patch posted by Timo Teräs to
ASTERISK-19106 for related dnsmgr issues.

(closes issue ASTERISK-19106)

Review: https://reviewboard.asterisk.org/r/1691/
........

Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 353397 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      de57235a
    • Alec L Davis's avatar
      Merged revisions 353369 via svnmerge from · f92d6412
      Alec L Davis authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r353369 | alecdavis | 2012-01-31 11:42:28 +1300 (Tue, 31 Jan 2012) | 9 lines
  
  Merged revisions 353368 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines
    
    prevent debug messsges displaying -ve Cseq numbers. Missed in R353320
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      f92d6412
    • Alec L Davis's avatar
      Merged revisions 353321 via svnmerge from · 0ccc1f52
      Alec L Davis authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines
  
  Merged revisions 353320 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines
    
    RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer
    
    * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers.
    
    * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
    
    Summary of CSeq numbers.
    An initial CSeq number must be less than 2^31
    A CSeq number can increase in value up to 2^32-1
    An incrementing CSeq number must not wrap around to 0.
    
    Tested with Asterisk 1.8.8.2 with Grandstream phones.
     
    alecdavis (license 585)
    Tested by: alecdavis
     
    Review: https://reviewboard.asterisk.org/r/1699/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0ccc1f52
    • Kevin P. Fleming's avatar
      Correct serious flaw in the top-level Makefile. · c6489d7b
      Kevin P. Fleming authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c6489d7b
    • Kevin P. Fleming's avatar
      Address OpenSSL initialization issues when using third-party libraries. · 92ef8a6f
      Kevin P. Fleming authored 
      When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.

This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.

Along the way, this patch also makes a few other minor changes:

* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
  more closely match what is used during run-time configuration.

* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
  instead of AC_PATH_PROG.

* Adds a new variable for linker flags in the build system (DYLINK), used for
  producing true shared libraries (as opposed to the dynamically loadable
  modules that the build system produces for 'regular' Asterisk modules).

* Moves the Makefile bits that handle installation and uninstallation of the
  main Asterisk binary into main/Makefile from the top-level Makefile.

* Moves a couple of useful preprocessor macros from optional_api.h to
  asterisk.h.

Review: https://reviewboard.asterisk.org/r/1006/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      92ef8a6f
    • Kevin P. Fleming's avatar
      Clarify log WARNING message when port-zero SDP 'm' lines received. · 82f313b7
      Kevin P. Fleming authored 
      Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
........

Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 353261 from http://svn.asterisk.org/svn/asterisk/branches/10


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      82f313b7
  19. Jan 29, 2012
  21. Jan 28, 2012
  23. Jan 27, 2012