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Review: https://reviewboard.asterisk.org/r/984/
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r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) | 10 lines
Analog 3-way call would not connect all parties if one was using sig_pri.
Also the "dahdi show channel" would not show the correct 3-way call
status.
* Synchronized the inthreeway flag between chan_dahdi and sig_analog.
* Fixed a my_set_linear_mode() sign error and made take an analog sub
channel enum.
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r293496 | pabelanger | 2010-11-01 12:09:05 -0400 (Mon, 01 Nov 2010) | 13 lines
Use ast_sockaddr_from_sin function not memcpy
This resolves some IAX2 registration issue report on the
asterisk-users mailing list.
(closes issue #18202)
Reported by: pabelanger
Patches:
update_registry.patch.v2 uploaded by pabelanger (license 224)
Tested by: pabelanger, Nic Colledge (mailing list)
Review: https://reviewboard.asterisk.org/r/993
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r293418 | rmudgett | 2010-10-29 20:53:29 -0500 (Fri, 29 Oct 2010) | 16 lines
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r293417 | rmudgett | 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) | 9 lines
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r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line
Remove some more code that serves no purpose.
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r293341 | rmudgett | 2010-10-29 19:46:41 -0500 (Fri, 29 Oct 2010) | 16 lines
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r293340 | rmudgett | 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) | 9 lines
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r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line
Remove some code that serves no purpose.
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r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) | 9 lines
Modify sip_setoption to not complain about unknown options.
This now behaves just like the other setoption callbacks. For the curious the
offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting
passed due to a fix for chan_local in 286189.
(closes issue #17985)
Reported by: globalnetinc
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r293197 | tilghman | 2010-10-28 15:00:06 -0500 (Thu, 28 Oct 2010) | 33 lines
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r293195 | tilghman | 2010-10-28 14:52:52 -0500 (Thu, 28 Oct 2010) | 12 lines
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r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines
"!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
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r293196 | tilghman | 2010-10-28 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines
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r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines
"!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
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r293159 | jpeeler | 2010-10-28 11:11:08 -0500 (Thu, 28 Oct 2010) | 18 lines
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r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines
Fix infinite loop in FILTER().
Specifically when you're using characters above \x7f or invalid character
escapes (e.g. \xgg).
(closes issue #18060)
Reported by: wdoekes
Patches:
issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717)
Tested by: wdoekes
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r293119 | jpeeler | 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) | 43 lines
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r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines
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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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r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) | 1 line
No need to define the struct if there are no users.
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r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) | 4 lines
Allow the DAHDI driver to compile, even with a sufficiently older version of libpri.
Fixes our Bamboo builds.
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r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) | 2 lines
Several more defines that need to be altered for compiling against an older version of libpri
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r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) | 4 lines
Allow the DAHDI driver to compile, even with a sufficiently older version of libpri.
Fixes our Bamboo builds.
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r292868 | dvossel | 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) | 39 lines
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r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines
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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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r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010) | 4 lines
Don't create directories without at least o+x
Also, making files that you are going to modify read-only is dumb.
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r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010) | 2 lines
Make files readable only by the owner
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r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines
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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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r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) | 45 lines
Add TLS cert helper script
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.
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r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct 2010) | 12 lines
Prevent multiple runs of event_sub_test from producing false failure results.
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.
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r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines
Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call.
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/
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r292667 | tilghman | 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010) | 2 lines
Compile correctly on Linux (asterisk/localtime.h depends upon asterisk/autoconfig.h loading first).
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r292628 | pabelanger | 2010-10-21 14:13:18 -0400 (Thu, 21 Oct 2010) | 5 lines
Fix typo in SUSE init script.
Reported by: Dave Cotton on asterisk-users list.
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r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) | 14 lines
Fixes recursive lock problem in manager.c
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
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r292557 | lmadsen | 2010-10-21 08:12:19 -0500 (Thu, 21 Oct 2010) | 14 lines
Merged revisions 292556 via svnmerge from
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r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) | 6 lines
Change res_ldap.sample.conf to match the schema.
(closes issue #17376)
Reported by: jcovert
Patches:
res_ldap.conf.sample.patch uploaded by jcovert (license 551)
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r292523 | russell | 2010-10-21 06:36:47 -0500 (Thu, 21 Oct 2010) | 4 lines
Add var=value to log message on update failure, and add newline.
... just for you, Leif.
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r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) | 7 lines
Send CONNECT_ACKNOWLEDGE for CIS calls too.
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.
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r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed, 20 Oct 2010) | 8 lines
Application not properly unregister in voicemail
(closes issue #18128)
Reported by: junky
Patches:
vm_unregister.diff uploaded by junky (license 177)
Tested by: pabelanger, lmadsen
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r292413 | pabelanger | 2010-10-20 20:07:17 -0400 (Wed, 20 Oct 2010) | 24 lines
Merged revisions 292412 via svnmerge from
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r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines
Merged revisions 292411 via svnmerge from
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r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines
Record priv-recordintro as sln, not gsm
This removes the gsm->sln step when transcoding
priv-recordintro.
(closes issue #18176)
Reported by: pabelanger
Patches:
chan_sip.diff uploaded by pabelanger (license 224)
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r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 Oct 2010) | 5 lines
Oops. This module uses the generic timer and no longer uses DAHDI.
This causes a problem with the Solaris and other system builds that have gcc
4.1 (where optional_api is non-optional).
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r292343 | pabelanger | 2010-10-19 18:14:23 -0400 (Tue, 19 Oct 2010) | 2 lines
Add resample and imap_tk dependencies.
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r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines
Add sip show peer info about crypto and remove dated comment
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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r292225 | lmadsen | 2010-10-18 16:51:23 -0500 (Mon, 18 Oct 2010) | 24 lines
Merged revisions 292224 via svnmerge from
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r292224 | lmadsen | 2010-10-18 16:50:47 -0500 (Mon, 18 Oct 2010) | 17 lines
Merged revisions 292222 via svnmerge from
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r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) | 9 lines
Add support for the new English (Australian Accent) sound files.
(closes issue #17426)
Reported by: camsown
Patches:
core-sounds-en_AU.txt uploaded by camsown (license 1050)
add_AU_sounds.patch.txt uploaded by lmadsen (license 10)
Tested by: camsown, lmadsen, jtodd, qwell
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r292227 | jpeeler | 2010-10-18 16:55:46 -0500 (Mon, 18 Oct 2010) | 25 lines
Merged revisions 292226 via svnmerge from
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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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r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) | 9 lines
Resolve some compiler errors in ast_sockaddr_is_any().
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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r292155 | dvossel | 2010-10-18 14:16:00 -0500 (Mon, 18 Oct 2010) | 2 lines
Fixes build error for systems not supporting IPV6_TCLASS.
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r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon, 18 Oct 2010) | 5 lines
Fix the cmgr parser.
(closes issue 0018152)
Reported by: menschentier
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r292085 | dvossel | 2010-10-18 11:02:17 -0500 (Mon, 18 Oct 2010) | 7 lines
Fixes qos settings for sockets bound to any IPv6 or IPv4 address.
(closes issue #18099)
Reported by: jamesnet
Patches:
issues_18099_v3.diff uploaded by dvossel (license 671
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r292083 | jpeeler | 2010-10-18 10:32:40 -0500 (Mon, 18 Oct 2010) | 4 lines
Disable use of inotify for call file handling as it is not working properly.
(related to #18089)
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r292050 | tzafrir | 2010-10-16 12:47:00 +0200 (ש', 16 אוק 2010) | 22 lines
Merged revisions 292049 via svnmerge from
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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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r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) | 5 lines
Ref/unref res_srtp when we create/destroy a session
This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp
tries to unload before chan_sip does. Thanks, Russell!
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r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) | 8 lines
Fixes peer's host port information being lost on sip reload.
(closes issue #18135)
Reported by: lmadsen
Patches:
crazy_ports_v2.diff uploaded by dvossel (license 671)
Tested by: lmadsen
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r291940 | pabelanger | 2010-10-15 15:50:22 -0400 (Fri, 15 Oct 2010) | 16 lines
Merged revisions 291939 via svnmerge from
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r291939 | pabelanger | 2010-10-15 15:35:20 -0400 (Fri, 15 Oct 2010) | 9 lines
Merged revisions 291938 via svnmerge from
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r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct 2010) | 2 lines
Clean up formatting.
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r291905 | twilson | 2010-10-15 09:39:58 -0700 (Fri, 15 Oct 2010) | 14 lines
Merged revisions 291904 via svnmerge from
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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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r291829 | dvossel | 2010-10-14 17:09:32 -0500 (Thu, 14 Oct 2010) | 8 lines
Set TCLASS field of IPv6 header when sip qos options are set.
(closes issue #18099)
Reported by: jamesnet
Patches:
issues_18099_v2.diff uploaded by dvossel (license 671)
Tested by: dvossel, jamesnet
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r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 Oct 2010) | 18 lines
Safer xml parsing, treat all clients the same, and better local candidate selection.
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.
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r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14 Oct 2010) | 10 lines
Add missing ifdefs for test framework and new locale code.
(closes issue #18137)
Reported by: ovi
Patches:
18137_test_framework_ifdef.patch uploaded by wdoekes (license 717)
18137_localelist_warning.patch uploaded by wdoekes (license 717)
Tested by: ovi
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r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines
Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
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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r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 Oct 2010) | 2 lines
Fix a typo - s/seucre/secure/
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