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r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines
Merged revisions 293004 via svnmerge from
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r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines
Fix inprocess_container in voicemail to correctly restrict max messages.
The comparison function logic was off, so the number of sessions for a given
mailbox were not being incremented properly. This problem caused the maximum
number of messages per folder to not be respected when simultaneously leaving
multiple voicemails just below the threshold.
These problems should be fixed by the above, but just in case:
Fixed resequence_mailbox to rely on the actual number of detected number of
files in a directory rather than just assuming only 10 messages more than the
maximum had been left. Also if more messages than the maximum are deleted they
are actually removed now.
The second purpose of this commit should have been separated out probably, but
is related to the above. Again, if the number of messages in a given voicemail
folder exceeds the maximum set limit make sure to allocate enough space for the
deleted and heard index tracking array.
A few random fixes:
There was a forgotten decrement of the inprocess count in imap_store_file.
When using IMAP storage, do not look in the directory where file based storage
messages may still reside and influence the message count.
Ensure to use only the first format in sendmail.
ABE-2516
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r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines
Merged revisions 292866 via svnmerge from
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r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines
This patch turns chan_local pvts into astobj2 objects.
chan_local does some dangerous things involving deadlock avoidance.
tech_pvt functions like hangup and queue_frame are provided with a
locked channel upon entry. Those functions are completely safe as
long as you don't attempt to give up that channel lock, but that is
impossible to guarantee due to the required deadlock avoidance necessary
to lock both the tech_pvt and both channels involved.
In the past, we have tried to account for this by doing things like
setting a "glare" flag that indicates what function should destroy the
pvt. This was used in local_hangup and local_queue_frame to decided
who should destroy the pvt if they collided in separate threads. I
have removed the need to do this by converting all chan_local tech_pvts
to astobj2. This means we can ref a pvt before deadlock avoidance
and not have to worry about that pvt possibly getting destroyed under
us. It also cleans up where we destroy the tech_pvt. The only unlink
from the tech_pvt container occurs in local_hangup now, which is where
it should occur.
Since there still may be thread collisions on some functions like
local_hangup after deadlock avoidance, I have added some checks to detect
those collisions and exit appropriately. I think this patch is going to
solve quite a bit of weirdness we have had with local channels in the past.
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r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines
Update the LDIF file for LDAP.
The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've
now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems
where I was doing an ldapadd to import the schema into the LDAP database, and the existing file
would cause problems and ERROR messages when registering.
Additional documention has been added based on feedback in the issue I'm closing.
(closes issue #13861)
Reported by: scramatte
Patches:
ldap-update.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jcovert, suretec, rgenthner
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The array of test subscriptions was declared "static," meaning that the
data.count field would retain its value between runs of the test. After the
first test run, this would result in false reports of test failures.
I chose to just remove the "static" keyword from the structure since it's not
a huge deal to construct this structure during each run of the test. Another
alternative would have been to zero out the data.count fields of each test
subscription instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This script is useful for quickly generating self-signed CA, server, and client
certificates for use with Asterisk. It is still recommended to obtain
certificates from a recognized Certificate Authority and to develop an
understanding how SSL certificates work. Real security is hard work.
OPTIONS:
-h Show this message
-m Type of cert "client" or "server". Defaults to server.
-f Config filename (openssl config file format)
-c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed)
-k CA key filename
-C Common name (cert field)
For a server cert, this should be the same address that clients
attempt to connect to. Usually this will be the Fully Qualified
Domain Name, but might be the IP of the server. For a CA or client
cert, it is merely informational. Make sure your certs have unique
common names.
-O Org name (cert field)
An informational string (company name)
-o Output filename base (defaults to asterisk)
-d Output directory (defaults to the current directory)
Example:
To create a CA and a server (pbx.mycompany.com) cert with output in /tmp:
ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
This will create a CA cert and key as well as asterisk.pem and the the two
files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem
and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem
and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will
require you to import the ca.crt file as a trusted cert.
To create a client cert using the CA cert created by the example above:
ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \
"My Company" -d /tmp -o joe_user
This will create client.crt/key/pem in /tmp. Use this if your device supports
a client certificate. Make sure that you have the ca.crt file set up as
a tlscafile in the necessary Asterisk configs. Make backups of all .key files
in case you need them later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a call is transfered by ECT or implicitly by disconnect in sig_pri or
implicitly by disconnect in chan_misdn, the connected line information is
not exchanged. The connected line interception macros also need to be
executed if defined.
The CALLER interception macro is executed for the held call.
The CALLEE interception macro is executed for the active/ringing call.
JIRA ABE-2589
JIRA SWP-2296
Patches:
abe_2589_c3bier.patch uploaded by rmudgett (license 664)
abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/958/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a AMI session to freeze because of invalid
use of recursive locks during the EVENT processing. This
patch removes the unnecessary locks.
(closes issue #18167)
Reported by: sustav
Patches:
manager_locking_v1.diff uploaded by dvossel (license 671)
Tested by: sustav
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The originator of the Q.SIG call completion signaling link was not changed
to the active state when the CONNECT message came in. The T309 processing
would immediately kill the signaling link because it was not in the active
state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds information about the encryption setting to 'sip show
peers' and removes an out-of-date comment from res_srtp.c and instead
directs users to the proper documentation.
(closes issue #18140)
Reported by: chodorenko
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r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines
Merged revisions 292223 via svnmerge from
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r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines
Fix improper operator key acceptance and clean up temp recording files.
This is a fix for when pressing the operator key after recording an unavailable,
busy, name, or temporary message in mailbox options. The operator key should not
be accepted here, but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or deleted as
apporopriate. Also, ensure removal of temporary recorded files after an early
hang up or when message acceptance confirmation times out.
ABE-2518
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These errors came up once this function was used from within netsock2.c.
The errors were like the following:
netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
The usage of a union here avoids this problem.
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r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines
Base directory for MOH should be ASTDATADIR
If the directive 'directory' is relative, make it relative to the
datadir, rather than to the varlibdir. In the sample configuration
it is relative ('moh').
This has no effect unless you have actively set the datadir explicitly
(at build time or at run time).
(closes issue #16906)
Patches:
moh_datadir uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/974/
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This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp
tries to unload before chan_sip does. Thanks, Russell!
(closes issue #18085)
Reported by: st
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r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines
Don't crash or deadlock on module unload
We can't hold the lock while pthread_join is called since aji_log_hook will
attempt to lock from the other therad. We reorder the pthread_join and
ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is
running, causing a crash.
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The gtalk channel driver was doing several unsafe operations
in regards to how it parsed incoming XML messages. I have cleaned
that code up so it should be much safer now.
We now treat all clients types the same. We have no reason to
distinguish between GMAIL and GOOGLE VOICE clients anymore because
they all work the same way.
I also modified how the local ip is found. If no bindaddress is provided
in the config file, we attempt to determine the local ip we
would use to connect to google.com. If that fails, then
we fall back to the ast_find_ourip() function as a last resort.
Using the new method makes it much less likely that we would ever
advertise a local RTP candidate as a loopback address.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While testing chan_gtalk I noticed jabber was using my IPv6 address
and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
to return both IPv6 and IPv4 results. Adding a family parameter gives you
the ablility to choose.
Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
Review: https://reviewboard.asterisk.org/r/973/
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r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines
Merged revisions 291643 via svnmerge from
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r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines
Deadlock between dahdi_exception() and dahdi_indicate().
There is a deadlock between dahdi_exception() and dahdi_indicate() for
analog ports. The call-waiting and three-way-calling feature can
experience deadlock if these features are trying to do something and an
event from the bridged channel happens at the same time.
Deadlock avoidance code added to obtain necessary channel locks before
attemting an operation with call-waiting and three-way-calling.
(closes issue #16847)
Reported by: shin-shoryuken
Patches:
issue_16847_v1.4.patch uploaded by rmudgett (license 664)
issue_16847_v1.6.2.patch uploaded by rmudgett (license 664)
issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/971/
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r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines
Merged revisions 291577 via svnmerge from
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r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines
Don't ignore frames that have been queued when softhangup'd
When an outgoing call is answered and hung up by the far end *very* quickly, we
may not read any frames and therefor end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately
sets the _softhangup flag on the channel and then queues the HANGUP control
frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates
that a hangup request has been made (which it will if _softhangup is set). So,
we end up losing control frames.
This change makes __ast_read continue to read frames even if a soft hangup has
been requested. It queues a hangup frame to make sure that __ast_read() will
still eventually return NULL.
Much thanks to David Vossel for all of the reviews, discussion, and help!
(closes issue #16946)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/740/
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This patch uses the XML docs to parse all of the available AMI commands
and allows you to enter the command name and be presented with a form with
the available fields. You can then rapidly tab through the fields and submit
the command and view the response. It is much faster/easier than having to
use telnet for testing purposes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The chan_dahdi faxdetect option only works for the first call. After that
the option no longer works. The struct dahdi_pvt.callprogress member is
the encoded user config setting for the callprogress and faxdetect config
options. Changing this value alters the configuration for all following
calls until the chan_dahdi.conf file is reloaded.
* Fixed the chan_dahdi ast_channel_setoption callback to not change the
users faxdetect config setting except for the current call.
* Fixed the chan_dahdi ast_channel_queryoption callback to read the active
DSP setting of the faxdetect option.
* Made actually disable the active faxdetect DSP setting for the current
call on the analog port. my_handle_dtmfup() is used for normal analog
ports. dahdi_handle_dtmfup() is the legacy code and is no longer used
unless in a radio mode.
(closes issue #18116)
Reported by: seandarcy
Patches:
issue18116_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/972/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines
Hold off ast_hangup() from destroying the ast_channel.
Must get the ast_channel lock before proceeding with release_chan() and
release_chan_early() to hold off ast_hangup() from destroying the
ast_channel.
Missed this change for -r291468.
JIRA ABE-2598
JIRA SWP-2317
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