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

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r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines

Fix 1.4 build when LOW_MEMORY is enabled.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A 'b' option has been added which causes chan_local to return the actual channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines

Be sure that we're not about to set bridgepvt NULL prior to dereferencing it.

(closes issue #11775)
Reported by: fujin


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12401)
 Reported by: davevg
 Patches: 
       astcli.diff2 uploaded by davevg (license 209)
 Tested by: davevg, Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines

Only try to prefix language if we are not using an absolute path (suffix it otherwise).

en/var/lib/asterisk/sounds/blah.gsm is a very silly path.

(closes issue #12379)
Reported by: kuj
Patches:
      12379-absolutepath.diff uploaded by qwell (license 4)
Tested by: kuj, qwell

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r114032 | file | 2008-04-10 12:58:39 -0300 (Thu, 10 Apr 2008) | 6 lines

Forgot the 1.4 branch for russian language fix.
(closes issue #12404)
Reported by: IgorG
Patches:
      voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)

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

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r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines

Create the directory where name recordings will go if it does not exist.
(closes issue #12311)
Reported by: rkeene
Patches:
      12311-mkdir.diff uploaded by qwell (license 4)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12404)
Reported by: IgorG
Patches:
      voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114027 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12409)
Reported by: candlerb


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines

Don't add custom URI options if they don't exist OR they are empty.
(closes issue #12407)
Reported by: homesick
Patches:
      uri_options-1.4.diff uploaded by homesick (license 91)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12396)
Reported by: tcalosi
Patches:
      12396.patch uploaded by putnopvut (license 60)
	  Tested by: tcalosi


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113980 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines

We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.

(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann

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r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines

If the [csv] section does not exist in cdr.conf, then an unload/load sequence
is needed to correct the problem.  Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113875 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #11901)
Reported by: pj


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were handled. In 1.4, if the absolute timeout were reached on a call, no matter what
the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T'
extension or hangup otherwise. The rearrangement of this function in trunk made this check
only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned
1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite
loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will
behave as they did in 1.4

(closes issue #11550)
Reported by: pj
Tested by: putnopvut



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113836 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12224)
Reported by: DEA
Patches:
      chan_skinny-debug-log.txt uploaded by DEA (license 3)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines

If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113752 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12254)
 Reported by: andrew
 Patches: 
       bug-12253.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines

If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.

(closes issue #12392)
Reported by: fnordian
Patches:
      chan_sip.patch uploaded by fnordian (license 110) with small modification from me


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113682 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #11954)
 Reported by: johan
 Patches: 
       app_dial_rev104031.patch uploaded by johan (license 334)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12390)
 Reported by: tzafrir
 Patches: 
       astcli_fixes.diff uploaded by tzafrir (license 46)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines

Initialize fr->cacheable to make valgrind happy

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(closes issue #12389)
 Reported by: davevg
 Patches: 
       astcli uploaded by davevg (license 209)


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r113507 | mmichelson | 2008-04-08 14:07:38 -0500 (Tue, 08 Apr 2008) | 8 lines

Fix potential buffer overflow that could happen if more than 100 announce files
were specified when calling ParkAndAnnounce. This overflow is not exploitable remotely
and so there is no need for a security advisory.

(closes issue #12386)
Reported by: davidw


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r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line

Add a little more that is required for previously added devices.
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines

Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975.

Thanks to Greg Oliver for providing me the required information.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113455 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12384)
Reported by: fnordian
Patches:
      features.patch uploaded by fnordian (license 110)

(patch modified by me, to give FEATURE_RETURN_* an AST_ prefix)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line

Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing.
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r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines

Add security note on astgenkey's manpage.
(closes issue #12373)
 Reported by: lmamane
 Patches: 
       20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)

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

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r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines

Move check for still-bridged channels out a little further, to avoid possible
deadlocks.  (Closes issue #12252)
Reported by: callguy
 Patches: 
       20080319__bug12252.diff.txt uploaded by Corydon76 (license 14)
 Tested by: callguy

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r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines

If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal

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https://origsvn.digium.com/svn/asterisk/trunk

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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines

Merged revisions 113012 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines

(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.



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

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r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines

Force ast_mktime() to check for DST, since strptime(3) does not.
(Closes issue #12374)

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

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r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines

Allow playback with noanswer (and add earlyrtp option).

(closes issue #9077)
Reported by: pj
Patches:
      earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn

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