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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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r291656 | rmudgett | 2010-10-13 18:45:11 -0500 (Wed, 13 Oct 2010) | 34 lines
Merged revisions 291655 via svnmerge from
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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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r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines
Merged revisions 291580 via svnmerge from
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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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r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13 Oct 2010) | 4 lines
More fixup for chan_gtalk.
This patch makes the xml parsing safer.
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r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010) | 8 lines
Add a simple AMI client web page
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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r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13 Oct 2010) | 26 lines
The chan_dahdi faxdetect option only works for the first FAX call.
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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r291507 | rmudgett | 2010-10-13 14:01:48 -0500 (Wed, 13 Oct 2010) | 18 lines
Merged revision 291504 from
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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r291469 | rmudgett | 2010-10-13 13:10:21 -0500 (Wed, 13 Oct 2010) | 23 lines
Merge revision 291468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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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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r291394 | russell | 2010-10-13 10:46:39 -0500 (Wed, 13 Oct 2010) | 20 lines
Merged revisions 291393 via svnmerge from
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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
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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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Add the extension and context of the calling channel to the log output if a macro could not be found.
(closes issue #18112)
Reported by: prado
Patches:
app_macro-info.diff uploaded by prado (license 510)
Tested by: schmidts
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r291284 | lmadsen | 2010-10-12 12:20:43 -0500 (Tue, 12 Oct 2010) | 15 lines
Merged revisions 291280 via svnmerge from
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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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r291265 | tilghman | 2010-10-12 12:06:23 -0500 (Tue, 12 Oct 2010) | 16 lines
Merged revisions 291264 via svnmerge from
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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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r291230 | lmadsen | 2010-10-12 11:08:04 -0500 (Tue, 12 Oct 2010) | 10 lines
Merged revisions 291229 via svnmerge from
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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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r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) | 16 lines
Fixes manager.c crash.
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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r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) | 2 lines
Update CHANGES to reflect new gtalk.conf options.
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r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
Gtalk enhancements and general code cleanup.
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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r291112 | rmudgett | 2010-10-11 13:48:15 -0500 (Mon, 11 Oct 2010) | 20 lines
Merged revisions 291110-291111 via svnmerge from
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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
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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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r291113 | rmudgett | 2010-10-11 13:51:13 -0500 (Mon, 11 Oct 2010) | 1 line
Move declaration closer to where now used.
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r291075 | rmudgett | 2010-10-11 11:42:54 -0500 (Mon, 11 Oct 2010) | 22 lines
Merged revisions 291073 via svnmerge from
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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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r291038 | tilghman | 2010-10-09 18:25:37 -0500 (Sat, 09 Oct 2010) | 2 lines
Add missing option to set calls to be logged in GMT/UTC.
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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
(closes 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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r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 Oct 2010) | 12 lines
Make outbound Google Voice calls.
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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MySQL should use.
(closes issue 17948)
Reported by qmax.
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r290864 | jpeeler | 2010-10-07 21:56:24 -0500 (Thu, 07 Oct 2010) | 23 lines
Merged revisions 290863 via svnmerge from
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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r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 Oct 2010) | 6 lines
Add Philippe Sultan to chan_gtalk author list.
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!
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r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 Oct 2010) | 5 lines
Outbound gtalk calls now work correctly.
There was a problem with how the candidates were being
built on an outbound call. This patch fixes that.
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r290752 | qwell | 2010-10-07 15:58:47 -0500 (Thu, 07 Oct 2010) | 23 lines
Merged revisions 290751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines
Allow PRI to build properly when using --with-pri.
Use the directories found for the parent when using lib dependencies.
(closes issue #17314)
Reported by: tzafrir
Patches:
17314-withdeps.diff uploaded by qwell (license 4)
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r290713 | russell | 2010-10-07 13:00:52 +0200 (Thu, 07 Oct 2010) | 11 lines
Merged revisions 290712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010) | 4 lines
Don't crash when Set() is called without a value.
Review: https://reviewboard.asterisk.org/r/949/
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r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 Oct 2010) | 4 lines
Fixes commented out code to use #if 0 instead.
Thanks to rmudgett for catching this!
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r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines
Fixes gtalk outbound DTMF to work properly.
Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered
this after investigating a packet capture from the gmail client. Instead of
performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive
on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue
with negotiating RTP payload type 106 for the telephony-event and then sending
DTMF as payload 101. This has been resolved by always negotiating 101 as the payload
type like we do everywhere else. With this patch, incoming google voice calls forwarded
to Asterisk via gtalk work.
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r290614 | rmudgett | 2010-10-06 13:50:37 -0500 (Wed, 06 Oct 2010) | 12 lines
Merged revision 290613 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines
Eliminate a redundant test for AST_CONTROL_REDIRECTING.
Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running
the redirecting interception macro if it is defined.
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r290576 | tilghman | 2010-10-06 08:49:19 -0500 (Wed, 06 Oct 2010) | 15 lines
Merged revisions 290575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) | 8 lines
Allow streaming audio from a pipe.
(closes issue #18001)
Reported by: jamicque
Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 Oct 2010) | 6 lines
Don't try to send RTP when remote_address is null
It is possible for ast_rtp_stop() to be called which will clear the remote
address and cause the sendto to fail and spam warnings. Don't send in this
case.
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r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05 Oct 2010) | 2 lines
Fixes uninitialized memory problem in 'iax2 set debug peer' option.
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r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines
Fixes chan_gtalk to work with gmail client
This patch was written by Philippe Sultan (phsultan). Thanks
for keeping this up to date!
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r290408 | tilghman | 2010-10-05 15:23:33 -0500 (Tue, 05 Oct 2010) | 22 lines
Merged revisions 290396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290396 | tilghman | 2010-10-05 15:21:02 -0500 (Tue, 05 Oct 2010) | 15 lines
Merged revisions 290392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) | 8 lines
Fix a crash by ensuring that we don't alter memory after it's freed.
(closes issue #17387)
Reported by: jmls
Patches:
20100726__issue17387.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
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r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05 Oct 2010) | 11 lines
Resolves dnsmgr memory corruption in chan_iax2.
(closes issue #17902)
Reported by: afried
Patches:
issue_17902.rev1.txt uploaded by russell (license 2)
Tested by: afried, russell, dvossel
Review: https://reviewboard.asterisk.org/r/965/
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r290376 | dvossel | 2010-10-05 14:56:29 -0500 (Tue, 05 Oct 2010) | 16 lines
Merged revisions 290375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines
Fixes PickupChan() not working with full channel name.
(closes issue #18011)
Reported by: schern
Patches:
app_directed_pickup.c.2.patch uploaded by schern (license 995)
app_directed_pickup.c.trunk.patch uploaded by schern (license 995)
Tested by: schern, dvossel
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r290289 | tilghman | 2010-10-05 09:15:46 -0500 (Tue, 05 Oct 2010) | 2 lines
Restore run directory for OS X, as well as standardizing some other paths to Mac OS X.
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