https://origsvn.digium.com/svn/asterisk/branches/1.4
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r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines
Fix DTMF not being sent to other side after a partial feature match
This fixes a regression from commit 176701. The issue was that
ast_generic_bridge never exited after the feature digit timeout had elapsed,
which prevented the queued DTMF from being sent to the other side.
This issue was reported to me directly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r191041 | seanbright | 2009-04-29 11:23:07 -0400 (Wed, 29 Apr 2009) | 6 lines
Fix a crash in app_queue with very long member lists.
A user reported via #asterisk that with very long lists of members, a crash
occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
of stack allocating copys of each interface name.
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r191422 | seanbright | 2009-05-01 11:42:48 -0400 (Fri, 01 May 2009) | 7 lines
Move the defintion of the a couple arrays out of loops.
According to Kevin, it is unspecified as to whether a variable defined inside
a block is allocated once by the compiler or for each pass through the block
(loops being the only interesting case), so just define these before we get
into our loop to be sure.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If backgrounding and no core will be produced, then changing the directory
won't break anything; likewise, if the CWD isn't accessible by the current
user, then a core wasn't possible anyway.
(closes issue #14831)
Reported by: chris-mac
Patches:
20090428__bug14831.diff.txt uploaded by tilghman (license 14)
20090430__bug14831.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For outgoing PTP redirected calls, you now need to use the inhibit(i)
option on all of the REDIRECTING statements before dialing the redirected-to
party. You still have to set the REDIRECTING(to-xxx,i) and the
REDIRECTING(from-xxx,i) values. The PTP call will update the redirecting-to
presentation when it becomes available and queue the redirecting update to
the calling channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r191096 | dbrooks | 2009-04-29 13:07:59 -0500 (Wed, 29 Apr 2009) | 8 lines
Patch to fix tab-completion crash on "remove extension"
This patch simply removes some old code back before Asterisk used editline.
This fixes the crash that occurred when tab-completing "remove extension".
(closes issue #14689)
Reported by: isaacgal
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I think it would behoove us to force "make validate-docs" to be run after the
XML documentation has been generated if dev-mode is enabled.
(closes issue #14989)
Reported by: tzafrir
Patches:
app_queue_xml.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Wait for a DivertingLegInformation3 message after receiving a
DivertingLegInformation1 message to complete the redirecting-to information
before queuing a redirecting update to the other channel.
* A DivertingLegInformation2 message should be responded to with a
DivertingLegInformation3 when the COLR is determined. If the call
could or does experience another redirection, you should manually
determine the COLR to send to the switch by setting REDIRECTING(to-pres)
to the COLR and setting REDIRECTING(to-num) = ${EXTEN}.
* A DivertingLegInformation2 message must have an original called number
if the redirection count is greater than one. Since Asterisk does
not keep track of this information, we can only indicate that the
number is not available due to interworking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines
Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
(closes AST-212)
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r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines
Fix a typo from 190661.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.
Review: http://reviewboard.digium.com/r/189
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit stops a warning message (user_data is NULL) from getting output when
manager events get sent before manager is initialized. This happens because manager
is initialized *after* modules are loaded and the act of loading modules triggers
manager events.
(issue #14974)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The messages sent by a technology when a connected line update is received
are best determined by the current call state of the channel. The struct
ast_party_connected_line.source value is really only useful as a possible
tracing aid.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r190187 | oej | 2009-04-23 12:07:26 +0200 (Tor, 23 Apr 2009) | 3 lines
unistd.h is required for usleep() on Darwin. It will not hurt to include it always
on other platforms either.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines
Fix a bug in chan_local glare hangup detection.
If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Jeff Peeler
Sean Bright
Joshua Colp
TransNexus OSP Development
Kevin P. Fleming
David Vossel
Richard Mudgett
David Brooks
Russell Bryant
Matthew Fredrickson
Mark Michelson
Olle Johansson