Age | Commit message (Collapse) | Author | Files | Lines |
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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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(closes issue #18128)
Reported by: junky
Patches:
vm_unregister.diff uploaded by junky (license 177)
Tested by: pabelanger, lmadsen
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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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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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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peers' and removes an out-of-date comment from res_srtp.c and instead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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r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010) | 3 lines
Fix typo in the sounds/Makefile.
(Issue #17426)
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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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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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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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(closes issue 0018152)
Reported by: menschentier
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(closes issue #18099)
Reported by: jamesnet
Patches:
issues_18099_v3.diff uploaded by dvossel (license 671
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(related to #18089)
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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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(closes issue #18135)
Reported by: lmadsen
Patches:
crazy_ports_v2.diff uploaded by dvossel (license 671)
Tested by: lmadsen
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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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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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(closes issue #18099)
Reported by: jamesnet
Patches:
issues_18099_v2.diff uploaded by dvossel (license 671)
Tested by: dvossel, jamesnet
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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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(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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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 makes the xml parsing safer.
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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.
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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/
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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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r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, 13 Oct 2010) | 16 lines
Memory overwrites when releasing mISDN call.
Phone <--> Asterisk
<-- ALERTING
--> DISCONNECT
<-- RELEASE
--> RELEASE_COMPLETE
* Add lock protection around channel list for find/add/delete operations.
* Protect misdn_hangup() from release_chan() and vise versa using the
release_lock.
JIRA ABE-2598
JIRA SWP-2317
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r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines
Merged revisions 291392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines
Lock pvt so pvt->owner can't disappear when queueing up a frame.
This fixes a crash due to a hangup race condition.
ABE-2601
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r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) | 7 lines
Add undocumented variables to phoneprov.conf.sample
(closes issue #18107)
Reported by: lathama
Patches:
phoneprov.conf.sample.diff uploaded by lathama (license 1028)
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r291264 | tilghman | 2010-10-12 12:05:31 -0500 (Tue, 12 Oct 2010) | 9 lines
Merged revisions 291263 via svnmerge from
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r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines
Oops, incorrect range (although unallocated at ARIN)
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r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) | 2 lines
Add documention that mentions options are defined but not used.
(Issue #18101)
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This issue was caused by improper use of the mansession lock and
manession_session lock. These two structures are confusing to begin
with so I'm not surprised this occurred. I fixed this by consistently
making sure we use each of these locks only to protect the data
in the corresponding structure. We had mismatched usage of these
locks which resulted in no mutual exclusivity occurring at all.
(closes issue #17994)
Reported by: vrban
Patches:
mansession_locking_fix.diff uploaded by dvossel (license 671)
Tested by: vrban
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This patch includes several chan_gtalk enhancements.
Two new gtalk.conf options have been added, externip
and stunadd. Setting externip allows us to
manually specify what the external IP address is
outside of a NAT environment. Setting the stunaddr
option to a valid stun server allows for that external
ip to be retrieved via a STUN server automatically. This
external IP is then advertised during call setup as
a possible candidate.
I have also attempted to clean up chan_gtalk's code
so it meets our coding guidelines. During this cleanup
I noticed several things that need to be done in the
code and made a TODO section at the top of the file.
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r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines
Merged revisions 291109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line
Add missing unlock to an exception condition in reload_config().
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r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line
Make exit from handle_request_do() consistent.
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r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines
Fixed infinite loop in verbose/debug message output.
Setting the module/filename specific message level and then changing it
resulted in the linked list being looped on itself. Traversing this
linked list is an infinite loop if what you are looking for is not in the
list.
Also plugged some CLI parsing holes in the associated CLI command:
* Removing a nonexistent module from the list actually added it with a
level of zero.
* Setting the non-module specific level to zero is now equivalent to
setting it to "off" as documented.
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Added options for faststart/h.245 tunneling per user/peer, properly
handle these and global options, correction of handling fs/tunneling
fields in signalling responses
(issue #17972)
Reported by: salecha
Patches:
fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
Tested by: may213, salecha
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This patch allows for outbound Google Voice calls to be
dialed from Asterisk using chan_gtalk. Below is an example
dialstring.
exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,)
In this example, 'asterisk' is the jabber.conf profile configured
to connect to your gmail account. In order to receive Google Voice
calls make sure to enable 'allowguest=yes' in gtalk.conf.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290938 f38db490-d61c-443f-a65b-d21fe96a405b
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MySQL should use.
(closes issue 17948)
Reported by qmax.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290937 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines
Ensure editline cleanup occurs when Ctrl-C is pressed at control console.
A recent change was made to avoid a race condition on shutdown which only called
the end functions from the console thread. However, when pressing Ctrl-C the
quit handler is called from the signal handler thread.
(closes issue #17698)
Reported by: jmls
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Philippe has made some notable contributions to the
gtalk channel driver. His name deserves to be listed
amoung the authors of that file. Thanks Philippe!
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290829 f38db490-d61c-443f-a65b-d21fe96a405b
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There was a problem with how the candidates were being
built on an outbound call. This patch fixes that.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290828 f38db490-d61c-443f-a65b-d21fe96a405b
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