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

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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111125 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines

update UPGRADE notes to document usage of the script

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

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r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines

This code change is made just for clarification. It does exactly
the same thing as before. It just doesn't look as wrong.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111123 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12025)
Reported by: agx


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111083 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines

Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.

(closes issue #10487)
Reported by: ewilhelmsen


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If this is wrong, I'd love to hear why.


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

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r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111019 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines

add a script to make getting the iLBC source code simple for end users

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Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.

2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.

3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.

4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.

5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.


(closes issue #11968)
Reported by: dimas
Patches:
      v2-11968-dsp.patch uploaded by dimas (license 88)
      v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines

If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be.
(closes issue #11995)
Reported by: fall

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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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to reload any file that references a given configuration file.


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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines

add note that the user will need to enable codec_ilbc to get it to build

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revert something dumb, because i was running svn diff in a subfolder not the root of trunk, before doing my commit and did not see it

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110930 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines

due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves

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change, it was possible to start Asterisk with the manager interface enabled, then
either comment out the enabled option or make manager.conf unopenable and the manager
interface would still be enabled.



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r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines

Make file access in cdr_custom similar to cdr_csv.

Fixes issue #12268.

Patch borrowed from r82344

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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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #11969)
 Reported by: pprindeville
 Patches: 
       acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12293)
 Reported by: pprindeville
 Patches: 
       bugid-0012293.1.6.patch uploaded by pprindeville (license 347)


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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


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Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12278)
Reported by: rjain
Patches: 
      chan_sip.c.diff uploaded by rjain (license 226)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue #12279)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk 
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue #12272)
Reported by: qq12345


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r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue #12280)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)


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Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.


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is subject to change while we work out the remaining issues.


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r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines

Don't attempt to do optimizations of gsm on mips platforms either.

(closes issue #12270)
Reported by: zandbelt
Patches:
      026-gsm-mips.patch uploaded by zandbelt (license 33)

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r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue #12266, reported by dimas)

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problems, but Qwell noticed the typo here.


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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

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other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue #12164, reported by milazzo and jsmith, patch by me)


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