[svn-commits] russell: trunk r275467 - /trunk/CHANGES
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Jul 10 09:48:07 CDT 2010
Author: russell
Date: Sat Jul 10 09:48:03 2010
New Revision: 275467
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=275467
Log:
Make indentation consistent, move some queue features to the queue section.
Modified:
trunk/CHANGES
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=275467&r1=275466&r2=275467
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Sat Jul 10 09:48:03 2010
@@ -93,12 +93,6 @@
------------
* Added 'p' option to PickupChan() to allow for picking up channel by the first
match to a partial channel name.
- * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
- timeout has expired.
- * Added 'R' option to app_queue. This option stops moh and indicates ringing
- to the caller when an Agent's phone is ringing. This can be used to indicate
- to the caller that their call is about to be picked up, which is nice when
- one has been on hold for an extened period of time.
* Added .m3u support for Mp3Player application.
* Added progress option to the app_dial D() option. When progress DTMF is
present, those values are sent immediately upon receiving a PROGRESS message
@@ -253,57 +247,63 @@
Queue changes
-------------
- * A new config option, penaltymemberslimit, has been added to queues.conf.
- When set this option will disregard penalty settings when a queue has too
- few members.
- * A new option, 'I' has been added to both app_queue and app_dial.
- By setting this option, Asterisk will not update the caller with
- connected line changes or redirecting party changes when they occur.
- * A 'relative-peroidic-announce' option has been added to queues.conf. When
- enabled, this option will cause periodic announce times to be calculated
- from the end of announcements rather than from the beginning.
- * The autopause option in queues.conf can be passed a new value, "all." The
- result is that if a member becomes auto-paused, he will be paused in all
- queues for which he is a member, not just the queue that failed to reach
- the member.
+ * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
+ timeout has expired.
+ * Added 'R' option to app_queue. This option stops moh and indicates ringing
+ to the caller when an Agent's phone is ringing. This can be used to indicate
+ to the caller that their call is about to be picked up, which is nice when
+ one has been on hold for an extened period of time.
+ * A new config option, penaltymemberslimit, has been added to queues.conf.
+ When set this option will disregard penalty settings when a queue has too
+ few members.
+ * A new option, 'I' has been added to both app_queue and app_dial.
+ By setting this option, Asterisk will not update the caller with
+ connected line changes or redirecting party changes when they occur.
+ * A 'relative-peroidic-announce' option has been added to queues.conf. When
+ enabled, this option will cause periodic announce times to be calculated
+ from the end of announcements rather than from the beginning.
+ * The autopause option in queues.conf can be passed a new value, "all." The
+ result is that if a member becomes auto-paused, he will be paused in all
+ queues for which he is a member, not just the queue that failed to reach
+ the member.
mISDN channel driver (chan_misdn) changes
----------------------------------------
- * Added display_connected parameter to misdn.conf to put a display string
- in the CONNECT message containing the connected name and/or number if
- the presentation setting permits it.
- * Added display_setup parameter to misdn.conf to put a display string
- in the SETUP message containing the caller name and/or number if the
- presentation setting permits it.
- * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
- indicate the dialplan settings are to be obtained from the asterisk
- channel.
- * Made misdn.conf parameter callerid accept the "name" <number> format
- used by the rest of the system.
- * Made use the nationalprefix and internationalprefix misdn.conf
- parameters to prefix any received number from the ISDN link if that
- number has the corresponding Type-Of-Number. NOTE: This includes
- comparing the incoming call's dialed number against the MSN list.
- * Added the following new parameters: unknownprefix, netspecificprefix,
- subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
- received number from the ISDN link if that number has the corresponding
- Type-Of-Number.
- * Added new dialplan application misdn_command which permits controlling
- the CCBS/CCNR functionality.
- * Added new dialplan function mISDN_CC which permits retrieval of various
- values from an active call completion record.
- * For PTP, you should manually send the COLR of the redirected-to party
- for an incomming redirected call if the incoming call could experience
- further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
- set the REDIRECTING(to-pres) to the COLR. A call has been redirected
- if the REDIRECTING(from-num) is not empty.
- * 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 (COLR) when it becomes available.
- * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
- information.
+ * Added display_connected parameter to misdn.conf to put a display string
+ in the CONNECT message containing the connected name and/or number if
+ the presentation setting permits it.
+ * Added display_setup parameter to misdn.conf to put a display string
+ in the SETUP message containing the caller name and/or number if the
+ presentation setting permits it.
+ * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
+ indicate the dialplan settings are to be obtained from the asterisk
+ channel.
+ * Made misdn.conf parameter callerid accept the "name" <number> format
+ used by the rest of the system.
+ * Made use the nationalprefix and internationalprefix misdn.conf
+ parameters to prefix any received number from the ISDN link if that
+ number has the corresponding Type-Of-Number. NOTE: This includes
+ comparing the incoming call's dialed number against the MSN list.
+ * Added the following new parameters: unknownprefix, netspecificprefix,
+ subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
+ received number from the ISDN link if that number has the corresponding
+ Type-Of-Number.
+ * Added new dialplan application misdn_command which permits controlling
+ the CCBS/CCNR functionality.
+ * Added new dialplan function mISDN_CC which permits retrieval of various
+ values from an active call completion record.
+ * For PTP, you should manually send the COLR of the redirected-to party
+ for an incomming redirected call if the incoming call could experience
+ further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
+ set the REDIRECTING(to-pres) to the COLR. A call has been redirected
+ if the REDIRECTING(from-num) is not empty.
+ * 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 (COLR) when it becomes available.
+ * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
+ information.
thirdparty mISDN enhancements
-----------------------------
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