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Make indentation consistent, move some queue features to the queue section.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
108
CHANGES
108
CHANGES
@@ -93,12 +93,6 @@ Applications
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------------
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------------
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* Added 'p' option to PickupChan() to allow for picking up channel by the first
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* Added 'p' option to PickupChan() to allow for picking up channel by the first
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match to a partial channel name.
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match to a partial channel name.
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* Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
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timeout has expired.
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* Added 'R' option to app_queue. This option stops moh and indicates ringing
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to the caller when an Agent's phone is ringing. This can be used to indicate
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to the caller that their call is about to be picked up, which is nice when
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one has been on hold for an extened period of time.
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* Added .m3u support for Mp3Player application.
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* Added .m3u support for Mp3Player application.
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* Added progress option to the app_dial D() option. When progress DTMF is
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* Added progress option to the app_dial D() option. When progress DTMF is
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present, those values are sent immediately upon receiving a PROGRESS message
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present, those values are sent immediately upon receiving a PROGRESS message
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@@ -253,57 +247,63 @@ Dialplan Variables
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Queue changes
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Queue changes
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-------------
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-------------
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* A new config option, penaltymemberslimit, has been added to queues.conf.
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* Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
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When set this option will disregard penalty settings when a queue has too
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timeout has expired.
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few members.
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* Added 'R' option to app_queue. This option stops moh and indicates ringing
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* A new option, 'I' has been added to both app_queue and app_dial.
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to the caller when an Agent's phone is ringing. This can be used to indicate
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By setting this option, Asterisk will not update the caller with
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to the caller that their call is about to be picked up, which is nice when
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connected line changes or redirecting party changes when they occur.
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one has been on hold for an extened period of time.
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* A 'relative-peroidic-announce' option has been added to queues.conf. When
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* A new config option, penaltymemberslimit, has been added to queues.conf.
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enabled, this option will cause periodic announce times to be calculated
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When set this option will disregard penalty settings when a queue has too
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from the end of announcements rather than from the beginning.
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few members.
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* The autopause option in queues.conf can be passed a new value, "all." The
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* A new option, 'I' has been added to both app_queue and app_dial.
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result is that if a member becomes auto-paused, he will be paused in all
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By setting this option, Asterisk will not update the caller with
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queues for which he is a member, not just the queue that failed to reach
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connected line changes or redirecting party changes when they occur.
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the member.
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* A 'relative-peroidic-announce' option has been added to queues.conf. When
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enabled, this option will cause periodic announce times to be calculated
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from the end of announcements rather than from the beginning.
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* The autopause option in queues.conf can be passed a new value, "all." The
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result is that if a member becomes auto-paused, he will be paused in all
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queues for which he is a member, not just the queue that failed to reach
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the member.
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mISDN channel driver (chan_misdn) changes
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mISDN channel driver (chan_misdn) changes
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----------------------------------------
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----------------------------------------
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* Added display_connected parameter to misdn.conf to put a display string
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* Added display_connected parameter to misdn.conf to put a display string
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in the CONNECT message containing the connected name and/or number if
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in the CONNECT message containing the connected name and/or number if
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the presentation setting permits it.
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the presentation setting permits it.
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* Added display_setup parameter to misdn.conf to put a display string
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* Added display_setup parameter to misdn.conf to put a display string
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in the SETUP message containing the caller name and/or number if the
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in the SETUP message containing the caller name and/or number if the
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presentation setting permits it.
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presentation setting permits it.
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* Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
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* Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
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indicate the dialplan settings are to be obtained from the asterisk
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indicate the dialplan settings are to be obtained from the asterisk
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channel.
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channel.
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* Made misdn.conf parameter callerid accept the "name" <number> format
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* Made misdn.conf parameter callerid accept the "name" <number> format
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used by the rest of the system.
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used by the rest of the system.
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* Made use the nationalprefix and internationalprefix misdn.conf
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* Made use the nationalprefix and internationalprefix misdn.conf
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parameters to prefix any received number from the ISDN link if that
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parameters to prefix any received number from the ISDN link if that
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number has the corresponding Type-Of-Number. NOTE: This includes
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number has the corresponding Type-Of-Number. NOTE: This includes
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comparing the incoming call's dialed number against the MSN list.
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comparing the incoming call's dialed number against the MSN list.
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* Added the following new parameters: unknownprefix, netspecificprefix,
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* Added the following new parameters: unknownprefix, netspecificprefix,
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subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
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subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
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received number from the ISDN link if that number has the corresponding
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received number from the ISDN link if that number has the corresponding
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Type-Of-Number.
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Type-Of-Number.
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* Added new dialplan application misdn_command which permits controlling
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* Added new dialplan application misdn_command which permits controlling
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the CCBS/CCNR functionality.
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the CCBS/CCNR functionality.
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* Added new dialplan function mISDN_CC which permits retrieval of various
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* Added new dialplan function mISDN_CC which permits retrieval of various
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values from an active call completion record.
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values from an active call completion record.
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* For PTP, you should manually send the COLR of the redirected-to party
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* For PTP, you should manually send the COLR of the redirected-to party
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for an incomming redirected call if the incoming call could experience
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for an incomming redirected call if the incoming call could experience
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further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
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further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
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set the REDIRECTING(to-pres) to the COLR. A call has been redirected
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set the REDIRECTING(to-pres) to the COLR. A call has been redirected
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if the REDIRECTING(from-num) is not empty.
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if the REDIRECTING(from-num) is not empty.
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* For outgoing PTP redirected calls, you now need to use the inhibit(i)
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* For outgoing PTP redirected calls, you now need to use the inhibit(i)
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option on all of the REDIRECTING statements before dialing the
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option on all of the REDIRECTING statements before dialing the
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redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
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redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
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and the REDIRECTING(from-xxx,i) values. The PTP call will update the
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and the REDIRECTING(from-xxx,i) values. The PTP call will update the
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redirecting-to presentation (COLR) when it becomes available.
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redirecting-to presentation (COLR) when it becomes available.
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* Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
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* Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
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information.
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information.
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thirdparty mISDN enhancements
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thirdparty mISDN enhancements
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-----------------------------
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-----------------------------
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