If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.
(closes issue #16006)
Reported by: viraptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010) | 11 lines
Fix regression for timed out parked call returning to caller
This issue seems to have been exposed by the fix in 160390 whereby using a
masquerade prevented a crash. The new channel used in the masquerade was
not copying the macro information from the old channel.
(closes issue #15459)
Reported by: djrodman
Patches:
patch_15459.txt uploaded by mnick (license )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The channel name comparison was not comparing the whole string and therefore
if one channel name was a substring of the other, the bridge would fail.
(closes issue #16528)
Reported by: telecos82
Patches:
res_features_r236843.diff uploaded by telecos82 (license 687)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_spawn_extension behaves differently from 1.4 in that hangups and extensions
that do not exist do not return an error, whereas in 1.6 it does. This is now
taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set
properly.
(closes issue #16106)
Reported by: ajohnson
Tested by: ajohnson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines
Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR.
(closes issue #14590)
Reported by: msetim
Patches:
queue_agent_userfield.patch uploaded by Laureano (license 265)
Tested by: Laureano, mnicholson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
and is set when a dynamic feature is triggered.
(closes issue #14663)
Reported by: tamiel
Patches:
20090313_features.diff uploaded by tamiel (license 712)
Tested by: tamiel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The find_channel_by_group callback was only looking at the channel that was
attempting to make the pickup instead of the other channels in the container.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Park() was not respecting the arguments passed to it. Any extension/context/priority
given to it was being ignored. This patch remedies this.
(closes issue #15380)
Reported by: DLNoah
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
Fix a case where CDR answer time could be before the start time involving parking.
(closes issue #13794)
Reported by: davidw
Patches:
13794.patch uploaded by murf (license 17)
13794.patch.160 uploaded by murf (license 17)
Tested by: murf, dbrooks
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines
If the "h" extension fails, give it another chance in main/pbx.c.
If the "h" extension fails, give it another chance in main/pbx.c, when it
returns from the bridge code. Fixes an issue where the "h" extension may
occasionally not fire, when a Dial is executed from a Macro.
Debugged in #asterisk with user tompaw.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines
Make ParkedCall application stop execution of the dialplan after hang up
Just changed park_exec to always return non-zero. I really wasn't entirely sure
at first if this was a bug. Decided it was since it would be surprising when
not using ParkedCall in the dialplan to hang up and have dialplan execution
continue.
(closes issue #14555)
Reported by: francesco_r
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The local struct ast_party_connected_line connected_caller variable
was uninitialized when the copy function was called.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines
Fix an incorrect assumption that certain values on the channel will always exist when they may not.
The CDR code involved with bridges wrongly assumed that the currently executing application and data
values will always exist. It is possible for this to be false when call forwarding is involved.
(closes issue #14984)
Reported by: gincantalupo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192462 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
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jeff Peeler
Tilghman Lesher
Sean Bright
Matthew Nicholson
Joshua Colp
David Vossel
Michiel van Baak
Russell Bryant
David Brooks
Kevin P. Fleming
Mark Michelson
Eliel C. Sardanons
Richard Mudgett