https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.8@290829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines
Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
