[asterisk-commits] kharwell: branch kharwell/pimp_sip_state r392023 - in /team/kharwell/pimp_sip...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jun 17 10:22:35 CDT 2013
Author: kharwell
Date: Mon Jun 17 10:22:34 2013
New Revision: 392023
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=392023
Log:
Multiple revisions 392004-392005
........
r392004 | mjordan | 2013-06-17 09:31:51 -0500 (Mon, 17 Jun 2013) | 5 lines
Restore bad merge on CHANGES
The patch for CDRs moved around a lot of content in CHANGES to try and
organize the areas that were affected. This missed some changes that went
in with a merge and removed some updates - this patch adds them back in.
........
r392005 | mjordan | 2013-06-17 09:40:23 -0500 (Mon, 17 Jun 2013) | 20 lines
Prevent sending a NewExten event after a Hangup during a stack restore
When a channel is originated, its application is typically set to AppDial2,
indicating that it was a dialed channel through the Dial API. Asterisk during
an originate will perform a stack execute to direct the outgoing channel to
a particular place in the dialplan or application. When the stack returns, the
previous application (AppDial2) is restored.
Unfortunately, in the case of an originated channel, the stack restore happens
after hangup. A stasis message is sent notifying everyone that the application
was restored, and this causes a NewExten event to go out after the Hangup event,
violating the basic contract consumers have of the channel lifetime. While we
could preclude the message from going out, restoring the channel's state before
it executed the next higher frame in the stack has to occur, and other places
in the code depend on this behavior.
Since we know that channel hung up (it's a ZOMBIE!), this patch simply checks
to see if the channel has been zombified before sending a NewExten event.
Note that this will fix a number of bouncing tests in the Test Suite. Go tests.
........
Merged revisions 392004-392005 from file:///srv/subversion/repos/asterisk/trunk
........
Merged revisions 392010 from http://svn.asterisk.org/svn/asterisk/team/group/pimp_my_sip
Modified:
team/kharwell/pimp_sip_state/ (props changed)
team/kharwell/pimp_sip_state/CHANGES
team/kharwell/pimp_sip_state/main/manager_channels.c
Propchange: team/kharwell/pimp_sip_state/
------------------------------------------------------------------------------
automerge = *
Propchange: team/kharwell/pimp_sip_state/
------------------------------------------------------------------------------
--- pimp_sip_state-integrated (original)
+++ pimp_sip_state-integrated Mon Jun 17 10:22:34 2013
@@ -1,1 +1,1 @@
-/team/group/pimp_my_sip:1-392006
+/team/group/pimp_my_sip:1-392020
Modified: team/kharwell/pimp_sip_state/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/kharwell/pimp_sip_state/CHANGES?view=diff&rev=392023&r1=392022&r2=392023
==============================================================================
--- team/kharwell/pimp_sip_state/CHANGES (original)
+++ team/kharwell/pimp_sip_state/CHANGES Mon Jun 17 10:22:34 2013
@@ -80,6 +80,11 @@
Reports 'InUse' for no logged in agents or no free agents.
Reports 'Idle' when an agent is free.
+ * The configuration options eventwhencalled and eventmemberstatus have been
+ removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
+ AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
+ sent. The "Variable" fields will also no longer exist on the Agent* events.
+
ResetCDR
------------------
* The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
@@ -92,6 +97,12 @@
------------------
* This application is deprecated in favor of the CHANNEL function.
+UserEvent
+------------------
+ * UserEvent will now handle duplicate keys by overwriting the previous value
+ assigned to the key. UserEvent invocations will also be distributed to any
+ interested res_stasis applications.
+
Core
------------------
@@ -99,6 +110,31 @@
that the REDIRECTING dialplan function can be used to set the redirecting
reason to any string. It also allows for custom strings to be read as the
redirecting reason from SIP Diversion headers.
+
+ * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
+ must be on the channel initiating the transfer to have any effect.
+
+ * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
+ driver specific. If the channel variable is set on the transferrer channel,
+ the sound will be played to the target of an attended transfer.
+
+ * The channel variable BRIDGEPEER becomes a comma separated list of peers in
+ a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
+ listed. Any more peers in the bridge will not be included in the list.
+ BRIDGEPEER is not valid in holding bridges like parking since those channels
+ do not talk to each other even though they are in a bridge.
+
+ * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
+ and will contain a value if the BRIDGEPEER's channel driver supports it.
+
+ * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
+ removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
+ activated the dynamic feature.
+
+ * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
+ only on the channel executing the dynamic feature. Executing a dynamic
+ feature on the bridge peer in a multi-party bridge will execute it on all
+ peers of the activating channel.
AMI (Asterisk Manager Interface)
------------------
@@ -257,6 +293,11 @@
The preferred way to configure parking is now through res_parking.conf while
configuration through features.conf is not currently supported.
+ * res_parking uses the configuration framework. If an invalid configuration is
+ supplied, res_parking will fail to load or fail to reload. Previously parking
+ lots that were misconfigured would generally be accepted with certain
+ configuration problems leading to individual disabled parking lots.
+
* Parked calls are now placed in bridges. This is a largely architectural change,
but it could have some implications in allowing for new parked call retrieval
methods and the contents of parking lots will be visible though certain bridge
@@ -296,6 +337,11 @@
by default. Instead, it will follow the timeout rules of the parking lot. The
old behavior can be reproduced by using the 'c' option.
+ * Added a channel variable PARKER_FLAT which stores the name of the extension
+ that would be used to come back to if comebacktoorigin was set to use. This can
+ be useful when comebacktoorigin is off if you still want to use the extensions
+ in the park-dial context that are generated to redial the parker on timeout.
+
Realtime
------------------
* Dynamic realtime tables for SIP Users can now include a 'path' field. This
@@ -310,6 +356,11 @@
* All future modules which utilize Sorcery for object persistence must have a
column named "id" within their schema when using the Sorcery realtime module.
This column must be able to contain a string of up to 128 characters in length.
+
+Security Events Framework
+-------------------------
+ * Security Event timestamps now use ISO 8601 formatted date/time instead of the
+ "seconds-microseconds" format that it was using previously.
Channel Drivers
@@ -401,22 +452,6 @@
If no resources exist or all are unavailable the device state is considered
to be unavailable.
-Security Events Framework
--------------------------
- * Security Event timestamps now use ISO 8601 formatted date/time instead of the
- "seconds-microseconds" format that it was using previously.
-
-Sorcery
-------------------
- * All future modules which utilize Sorcery for object persistence must have a
- column named "id" within their schema when using the Sorcery realtime module.
- This column must be able to contain a string of up to 128 characters in length.
-
-app_userevent
-------------------
- * UserEvent will now handle duplicate keys by overwriting the previous value
- assigned to the key. UserEvent invocations will also be distributed to any
- interested res_stasis applications.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
Modified: team/kharwell/pimp_sip_state/main/manager_channels.c
URL: http://svnview.digium.com/svn/asterisk/team/kharwell/pimp_sip_state/main/manager_channels.c?view=diff&rev=392023&r1=392022&r2=392023
==============================================================================
--- team/kharwell/pimp_sip_state/main/manager_channels.c (original)
+++ team/kharwell/pimp_sip_state/main/manager_channels.c Mon Jun 17 10:22:34 2013
@@ -638,6 +638,11 @@
return NULL;
}
+ /* Ignore any updates if we're hungup */
+ if (ast_test_flag(&new_snapshot->flags, AST_FLAG_ZOMBIE)) {
+ return NULL;
+ }
+
if (old_snapshot && ast_channel_snapshot_cep_equal(old_snapshot, new_snapshot)
&& !strcmp(old_snapshot->appl, new_snapshot->appl)) {
return NULL;
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