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features that will always be present in DAHDI
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locked.
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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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more clear
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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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dedicated
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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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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backwards,
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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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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the config sample file. Also fix "sip show settings" for tcp connections.
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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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r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines
Zap/pseudo is ten characters, but DAHDI/pseudo is
twelve. The strncmp call in next_channel should
account for this.
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r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines
Update the "last" channel in next_channel in app_chanspy so
that the same pseudo channel isn't constantly returned.
related to issue #13124
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r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
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(closes issue #13137)
Reported by: eliel
Patches:
chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64)
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(closes issue #13085)
Reported by: pabelanger
Patches:
logrotate uploaded by pabelanger (license 224)
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r132974 | mmichelson | 2008-07-23 12:05:04 -0500 (Wed, 23 Jul 2008) | 4 lines
Fix building of chan_dahdi when HAVE_PRI is not
defined.
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r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines
minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool)
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r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line
another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
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pointer...
Thanks, eliel!
Reported by: eliel
Patches:
chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
(closes issue #13135)
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https://origsvn.digium.com/svn/asterisk/trunk
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Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
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as libpri was installed and enabled
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