If no number is specified in the SIP_HEADER dialplan function, then just use the first one. (issue #7854 reported by sxpert and issue #7863 reported by hristo)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

along that line.



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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it 
had a non static function when it should.


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having a bunch of erroneous messages about attempting to double free frames
spammed to the console.  Problem reported to me by file ...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ignore DTMF begin frames in the waitstream core so that we don't get duplicate digits in our extension match


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41545 65c4cc65-6c06-0410-ace0-fbb531ad65f3

define it to provide a result of "0" and change the code appropriately.
(issue #7805)


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Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update configure script to check for Zaptel VLDTMF support since this will now be required. If you build and don't have Zap stuff, upgrade your zaptel copy. As well - change some minor things so that we get "checking for" and not "checking checking for".


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a resolution to that bug report)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

does not need special code in chan_sip any more.


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........
r41411 | russell | 2006-08-30 14:59:44 -0400 (Wed, 30 Aug 2006) | 6 lines

Restore original functionality of 1.2 in places where ANI was not set, but was
changed to be set.  The original change was done to ensure that the behavior of
the "callerid" option in each channel driver was consistent, but it caused an
unexpected behavior change of CDR records for users, so this change is being
reverted in 1.2.  (issue #7695)

........


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https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r41390 | file | 2006-08-30 13:58:31 -0400 (Wed, 30 Aug 2006) | 2 lines

Properly handle an ETIMEDOUT result from pthread_cond_timedwait (issue #7318 reported by arkadia)

........


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translated frame to the spies, *not* the original frame.  Thanks PCadach!


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Print what read-only variable is being set so that the user doesn't have to guess and figure it out or look at the list in cdr.c and match it up (issue #7831 reported by sedwards)


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Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking


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If the RTP stack is already being operated in callback mode, then suspend it upon switching to P2P callback bridging. Once P2P callback bridging has ended, then restore callback mode.


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