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
certified/1.8.6
Russell Bryant 15 years ago
parent 405d6cdf31
commit fcaac09507

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