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

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r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul 2008) | 2 lines

use the proper method for building version.h

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respectively.  Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
 Reported by: eliel
 Patches: 
       cliprompt.patch uploaded by eliel (license 64)


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(closes issue #12974)
 Reported by: jcollie
 Patches: 
       0001-Don-t-override-duplicate-optimization-flags.patch uploaded by jcollie (license 412)


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build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI


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

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r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines

Detect when sox fails to raise the volume, because sox can't read the file.
(closes issue #12939)
 Reported by: rickbradley
 Patches: 
       20080728__bug12939.diff.txt uploaded by Corydon76 (license 14)
 Tested by: rickbradley

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implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.

On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.

Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.

closes issue #11928)
Reported by: adriavidal
Patches:
      1.6.0-configurev2.patch uploaded by putnopvut (license 60)



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(closes issue #13134)
 Reported by: eliel
 Patches: 
       app_image.c.patch uploaded by eliel (license 64)
       UPGRADE.patch uploaded by eliel (license 64)


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(closes issue #13176)
Reported by: pj


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called from elsewhere in Asterisk to find the current state of a device.  In
that case, we want to use the cached value if it exists.  The other way is when
processing a device state change.  In that case, we do not want to check the
cache because returning the last known state is counter productive.


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It is no longer needed.


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(closes issue #13155)
 Reported by: greenfieldtech
 Patches: 
       app_voicemail.c.patch uploaded by greenfieldtech (license 369)
       hebrew.ods uploaded by greenfieldtech (license 369)


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(closes issue #13163)
 Reported by: suretec
 Patches: 
       asterisk.ldif uploaded by suretec (license 70)


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we do NOT need to uri_decode in manager.
(if I sent core%20show%20channels from a telnet
session, it should be interpreted literally, however,
if I send that from an http session, it should be
decoded, which is the behaivor now)


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r133709 | tilghman | 2008-07-25 13:00:33 -0500 (Fri, 25 Jul 2008) | 3 lines

Remove unnecessary mmap flag
(Closes issue #13161)

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

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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines

Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
 Reported by: davidw
 Patches: 
       20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
 Tested by: davidw, jvandal, murf

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ago (does not affect 1.4), where you would pass
a pointer to the end of a character array, and
ast_uri_decode would do no good.


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

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r133578 | russell | 2008-07-25 10:00:31 -0500 (Fri, 25 Jul 2008) | 10 lines

Merged revisions 133577 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) | 2 lines

Fix the IAX2 URI for calling Digium

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to Asterisk licensing information.  The licensing page includes the Asterisk license,
as well as a (not yet complete) list of 3rd party libraries that may be used, as well
as what license we receive them under.

Help filling out this list in the format that I have started in doxyref.h would be
much appreciated.  :)


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

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r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines

We need to make sure to null-terminate the "name"
portion of SIP URI parameters so that there are no
bogus comparisons.

Thanks to bbryant for pointing this out.


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Pointed out by Atis Lezdins in #asterisk-dev



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 - Use ast_strlen_zero in one place
 - check for successful string comparison the way most of Asterisk code does it


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the code checks to see if there is a cached state available, use the aggregate
cached state across all servers, and not just the local state.


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

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r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines

Fix rtautoclear and rtcachefriends
(Closes issue #12707)

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this is the right event to subscribe to ...


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this commit, only the logger thread's PID would
be printed.

(closes issue #13150)
Reported by: atis
Patches:
      log_pid.diff uploaded by putnopvut (license 60)
Tested by: eliel




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(Closes issue #13147)


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(Closes issue #13148)


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Reported by: murf
Tested by: murf
For: J. Geis

The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.

So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.

I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.

I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see 
http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html




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

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r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul 2008) | 1 line

inbandrelease is gone - it's now inbanddisconnect
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r133237 | kpfleming | 2008-07-23 16:05:22 -0500 (Wed, 23 Jul 2008) | 2 lines

revert an optimization that broke ABI... thanks russell!

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Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections.


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

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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines

As suggested by seanbright, the PSEUDO_CHAN_LEN in 
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.

Also changed the next_unique_id_to_use to have the 
static qualifier.

Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.


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