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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.26@212958 f38db490-d6...v1.4.26kpfleming8-20/+19
2009-07-20Importing release summary for 1.4.26 release.lmadsen2-0/+1482
2009-07-20Update .version and ChangeLog files.lmadsen2-1/+5
2009-07-20Create release of 1.4.26 from tag 1.4.26-rc6.lmadsen0-0/+0
2009-07-13Use autotagged externalslmadsen0-0/+0
2009-07-13Importing files for 1.4.26-rc6 release.lmadsen3-0/+25255
2009-07-13Creating tag for the release of asterisk-1.4.26-rc6lmadsen0-0/+0
2009-07-13Print CID match in "show dialplan".russell1-1/+4
2009-07-10Properly ACK 487 responses to canceled INVITEs.mmichelson1-3/+9
2009-07-10SIP registration auth loop caused by stale noncedvossel1-7/+25
2009-07-10Ensure that outbound NOTIFY requests are properly routed through stateful pro...mmichelson1-2/+6
2009-07-09No audio on calls from Asterisk to various ISDN devices until DTMF sent by ca...rmudgett1-0/+1
2009-07-09Changing ast_samp2tv to not use floating point.dvossel1-1/+1
2009-07-08Fixes 8khz assumptionsdvossel3-29/+33
2009-07-08moving ast_devstate_to_extenstate to pbx.c from devicestate.cdvossel4-34/+34
2009-07-08Prevent phantom calls to queue members.mmichelson1-2/+3
2009-07-08Update config.guess and config.sub from the savannah.gnu.org git repo.qwell2-103/+225
2009-07-08ast_samp2tv needs floating point for 16khz audiodvossel1-1/+1
2009-07-08Add redirection warnings for the invalid language codes previously removed.tilghman1-0/+72
2009-07-08Make OpenSSL usage thread-safe.russell1-1/+45
2009-07-02Removed confusing warning message "Got Busy in Connected State"rmudgett1-2/+0
2009-07-02moving device state functions from pbx.h to devicestate.h to sync with other ...dvossel4-171/+171
2009-07-02Improved mapping of extension states from combined device states.dvossel4-92/+232
2009-06-30More incorrect language codes, plus ensuring that regionalizations use the sp...tilghman2-189/+191
2009-06-30Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in p...qwell1-2/+2
2009-06-30"tw" is the language specification for Twi (from Ghana) not Taiwanese.tilghman2-37/+42
2009-06-29Add error message so that it is clear why a SIP peer was not processed whenmmichelson1-0/+1
2009-06-29Fix build oops.mmichelson1-1/+1
2009-06-29Fix a problem where chan_sip would ignore "old" but valid responses.mmichelson1-44/+44
2009-06-29Revision 189537 was supposed to make 1.4 more correct. Instead, it broke fun...tilghman2-35/+2
2009-06-29segfault after SPINLOCK schedule deletedvossel1-4/+14
2009-06-29Place unlock of mutex in an else block so that it does not get unlocked twice.mmichelson1-1/+2
2009-06-27The ISDN CPE side should not exclusively pick B channels normally.rmudgett1-8/+5
2009-06-26Make sure to recreate the dahdi pseudo channel after dahdi restartjpeeler1-1/+1
2009-06-26Don't fast forward past the end of a message.russell1-0/+7
2009-06-26Fixing voicemail's error in checking max silence vs min message lengthdbrooks1-1/+1
2009-06-25I didn't see that Mark already fixed the underlying issue!twilson1-3/+2
2009-06-25Fix a case where CDR answer time could be before the start time involving par...russell1-1/+10
2009-06-25Don't try to free NULLtwilson1-1/+3
2009-06-25Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.mmichelson1-1/+7
2009-06-25Resolve a crash related to a T.38 reinvite race condition.russell1-16/+20
2009-06-24Improved chan_dahdi.conf pritimer error checking.rmudgett1-15/+26
2009-06-24Use the handy UNLINK macro instead of hand-coding the same thing in-line.mmichelson1-5/+1
2009-06-23MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and ...dvossel1-1/+9
2009-06-23Fix more memory leaks that may result if rtp is not successfully allocated.mmichelson1-0/+4
2009-06-23Fix potential memory leak in chan_sip when video rtp is not allocated properly.mmichelson1-0/+4
2009-06-22Report CallerID change during a masquerade.russell1-15/+25
2009-06-22Make Polycom subscription type override check more explicit.russell1-5/+9
2009-06-22Remove an extra debug line left from previous commit.mmichelson1-1/+0
2009-06-22Fix a situation in which Asterisk would not stop retransmitting 487s.mmichelson1-2/+34
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