Multiple Schedules and Auto Attendant in UC560
Dear all,
I have configured two schedules in UC560 using CCA 3.0.1 (office_hours and break_time). I have then configured two auto-attendants. AA1 is used for office_hours and AA2 for break_time. AA1 is working fine. However, when it is break time, AA2 is not being used, instead the menu for closed hours in AA1 is used. If I swap office_hours and break_time as AA2 and AA1, then break_time is ok while office_hours settings do not work.
Internally, I can dial both AA extension numbers and get the correct prompt. The problem is when an external call comes in.
Any ideas what I may be doing wrong?
Hi Tiziana,
Sorry, I should have been more specific. What you have there is the built in script editor express. You can download the standalone CUE script editor here:
http://www.cisco.com/cisco/software/release.html?mdfid=282825559&softwareid=282774364&release=7.0.6&relind=AVAILABLE&rellifecycle=&reltype=latest
Edit to add: I think there are also some sample scripts that may help you out there as well. Here is also the Unity Express Script Editor guide.
Best,
David
Similar Messages
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For some reason I'm only able to get the AA to work in PBX mode. If I try Blend or Key System, when someone calls in, the line rings it seems like the AA picks up but I get silence. I would like to use the Key system config.
Hi James,
Please send your configuration through Configuration Utility -> Services -> Feedback, input "Attention [email protected]" at Issues or suggestions section. I will take a look and get back to you.
Best regards,
Wendy -
New UM Cert only half works? Lost voice mail, kept Auto Attendant
We have a Lync 2013 environment with a collocated front end/mediation server using Exchange 2010 for UM services (voice mail and auto attendant to pick up and transfer by extension) Today our UM cert expired so I replaced it with a new one from a new internal
CA we brought up. Exchange accepted the cert (its valid for use) and applied the UM service to it with no problems. After restarting the UM service and testing I have noticed that we have lost voice mail since the switch.
In the past whenever we had a problem with the UM cert or UM in general we have lost access to BOTH voice mail and our auto attendant so this is weird to me. I noticed I am getting my missed call notifications in outlook, but no voicemail (and not being
prompted to leave one as the calling party either).
Event viewer isn't showing much for errors. Here is one that sounds related, but it doesn't appear every time I try to make a test call to voice mail.
The Unified Messaging server encountered an error while trying to process the message with header file "C:\Program Files\Microsoft\Exchange Server\V14\UnifiedMessaging\voicemail\12635b29-9d7f-4e2f-a274-e80a4c4bc04c.txt". Error details: "Microsoft.Exchange.UM.UMCore.SmtpSubmissionException:
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commandHi,
Please go to the voicemail folder under \Program Files\Microsoft\Exchange Server\V14\UnifiedMessaging\ to check if the folder is empty.
Please check the sentence below in the link:
http://blogs.technet.com/b/kamehta/archive/2010/12/20/um-2010-voicemail-delivery-failure-to-hub-transport.aspx
“If the folder is empty, then the voicemail files have left the UM and on its way to Hub Transport.
If you see voicemail audio files there it means the connection between UM and HT is broken somewhere. This can arise because of one of the following scenarios.
The HT receive connector is mis-configured. Check the AUTH on the receive connectors
Verify if you can telnet on hub transport's IP and port number 25
Verify if HT FQDN is resolving to the correct IP
Check if any anti-virus software is installed”
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
We have two companies on the sam CME/CUE system. Is there a way to setup two Auto Attendants and have a single button push to turn on night server for each auto attendant.
We have one person who answers the phones for both companies during the day, but after hours we turn on the auto attendant with a night service button. This works fine for the single company, but now that we are starting to recieve calls for the second company after hourse... we'd like to take advantage of a second Auto Attendant with that night service.Hi Darren,
This is supported :)
Cisco CME 3.2.1 and later versions support the creation of multiple AA services that feed into a single call-queue service that manages up to ten ephone hunt groups (individual call queues). Each of the AAs can be set up to use different options or to reach different hunt groups, and AAs can also share hunt groups. For instance, you can have three AAs that each use three hunt groups, or you can have five AAs that share some of the ten hunt groups, or ten AAs that each use one hunt group. This flexibility allows companies to create different automatic-attendant treatment for different classes of callers.
For example, you can set up an AA in interactive mode to answer calls using a prerecorded message that offers various menu choices to callers. One type of menu choice is to allow a caller to press a digit to be connected to a department or service (hunt group). Another type of menu choice can allow the caller to dial a known extension number directly.
Alternatively, you can set up an AA in drop-through mode, a new feature with Cisco CME 3.3, which is described in the "Drop-Through Mode" section. An AA that is set up in drop-through mode transfers incoming calls directly into a call queue for a hunt group without allowing any interactive choice by the caller. A prompt is optional in drop-through mode. When you configure multiple AAs, each AA can be independently assigned to interactive or drop-through mode.
When you set up multiple interactive AAs, separate welcome prompts must be recorded for each AA. With multiple AAs, the welcome prompt is used to inform callers about the menu choices that are available to them.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html
Hope this helps!
Rob -
Cisco UC560 java.lang.NullPointerException error when opening "auto attendant" CCA
Hi Guys,
In CCA i get the error:java.lang.NullPointerException and it is only when selecting "Auto Attendant".
No Configuration has been done via CLI
Cisco IOS Software, UC500 Software (UC500-ADVIPSERVICESK9-M), Version 15.1(4)M5, RELEASE SOFTWARE (fc1)
CCA version 3.2(2)
Attaching CCA logs
Any Ideas ?I would suggest downloading the latest version of the IDE (v. 5.0) which is available at:
http://wwws.sun.com/software/sundev/jde/index.html -
I need a multiple message voice mail app for Iphone 4S that also provides an auto-attendant, e.g the caller can dial 1 for one of business's or 2 for the other. Each of these options will need to have a different voice mail greeting. Help please?
There are no alternative voicemail apps - the core functionality of the phone can not be replaced.
You'll have to look for an external service, that can then forward calls for each caller onto the correct phone.
At our business we use Voipfone.co.uk which allows multiple phone lines to come into one VOIP account. -
Exchange 2013 Auto Attendant fails when using multiple languages
Hi All,
Trying to setup UM Auto Attendants as I had it in Exchange 2010, but cannot get it to work as I want.
Here is my setup:
Windows 2012 and Exchange 2013.
Parent - Main Auto Attendant (80)
Child - English Auto Attendant (81)
Child - Swedish Auto Attendant (82)
Child - Finnish Auto Attendant (83)
All callers are first connected to 80 where they can select the language to continue with (press 81 for English, press 82 for Swedish, press 83 for Finnish).
This is what I found out this far:
- If I have English language selected on all Attendants everything works fine I can select all choices and everything works as it should.
- If I select the proper language for the other languages (82 & 83) only the connection to 80 & 81 works. If I try to connect from 80 to 82 or 83 nothing happens (connection goes quiet) and the only solution to get UM to answer again is to restart the
UM service on the server.
Extra info/what I tested this far:
- I reinstalled the extra language packs just to be sure there were no issues.
- If I select Swedish or Finnish language on extension 80 they work fine, so the language packs seems to be OK.
Ideas Anyone?
//AseHi All,
After installing CU1 for Exchange 2013 this behaviour changed a little bit. Instead of getting total freeze of the system I get an error message instead saying "System Error Occured" and then a hangup.
I also find out that if I remove the checkbox "Set the auto attendant to respond to voice commands" for the extension 82 & 83 (Swedish & Finnish Auto Attendants) this error does not happen, but I like to use the voice commands....
Also when the error occurs I get the following in the Event Log:
The VoIP platform encountered an exception Microsoft.Exchange.UM.UMCommon.UMGrayException: A non-fatal exception occurred. For details, please see the inner exception. ---> System.InvalidOperationException: The language for the grammar does not match
the language of the speech recognizer.
at Microsoft.Speech.Recognition.RecognizerBase.ThrowIfSapiErrorCode(SAPIErrorCodes errorCode)
at Microsoft.Speech.Recognition.FileGrammarContent.Load(SapiGrammar sapiGrammar, Boolean enabled, Single weight, Int32 priority)
at Microsoft.Speech.Recognition.GrammarContent.Load(SapiGrammar sapiGrammar)
at Microsoft.Speech.Recognition.Grammar.Load(SapiGrammar sapiGrammar, IRecognizerInternal recognizer)
at Microsoft.Speech.Recognition.RecognizerBase.LoadGrammarIntoSapi(Grammar grammar)
at Microsoft.Speech.Recognition.RecognizerBase.LoadGrammar(Grammar grammar)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.PlayPromptsAndRecoSpeechSessionState.LoadGrammar(UMGrammar grammar)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.PlayPromptsAndRecoSpeechSessionState.LoadGrammars()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.PlayPromptsAndRecoSpeechSessionState.InternalStart()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.SessionState.Start()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.ChangeState(SessionState nextState)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.PlayPrompts(ArrayList prompts, Int32 minDigits, Int32 maxDigits, Int32 timeout, String stopTones, Int32 interDigitTimeout, StopPatterns stopPatterns, Int32 startIdx, TimeSpan offset, List`1
grammars, Boolean expetingSpeechInput, Int32 babbleTimeout, Boolean stopPromptOnBargeIn, String turnName, Int32 initialSilenceTimeout)
at Microsoft.Exchange.UM.UMCore.SpeechMenu.PlayPrompts(ArrayList prompts, BaseUMCallSession vo)
at Microsoft.Exchange.UM.UMCore.SpeechMenu.SpeechMenuStart(BaseUMCallSession vo)
at Microsoft.Exchange.UM.UMCore.SpeechMenu.StartActivity(BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.ActivityManager.ChangeActivity(ActivityBase next, BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.TransitionBase.Execute(ActivityManager manager, BaseUMCallSession vo)
at Microsoft.Exchange.UM.UMCore.Menu.OnComplete(BaseUMCallSession vo, UMCallSessionEventArgs voiceObjectEventArgs)
at Microsoft.Exchange.UM.UMCore.Menu.StartActivity(BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.ActivityManager.ChangeActivity(ActivityBase next, BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.TransitionBase.Execute(ActivityManager manager, BaseUMCallSession vo)
at Microsoft.Exchange.UM.UMCore.Menu.OnComplete(BaseUMCallSession vo, UMCallSessionEventArgs voiceObjectEventArgs)
at Microsoft.Exchange.UM.UMCore.Menu.StartActivity(BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.ActivityManager.ChangeActivity(ActivityBase next, BaseUMCallSession vo, String refInfo)
at Microsoft.Exchange.UM.UMCore.ActivityBase.OnHeavyBlockingOperation(BaseUMCallSession vo, HeavyBlockingOperationEventArgs hboea)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.HeavyBlockingOperationSessionState.CompleteFinalAsyncCallback()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.SessionState.CompleteNonTeardownState()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.<>c__DisplayClassf.<CatchAndFireOnError>b__b()
at Microsoft.Exchange.Common.IL.ILUtil.DoTryFilterCatch(TryDelegate tryDelegate, FilterDelegate filterDelegate, CatchDelegate catchDelegate)
--- End of inner exception stack trace ---
at Microsoft.Exchange.UM.UMCommon.ExceptionHandling.ExceptionCatcher(Object exception)
at Microsoft.Exchange.Common.IL.ILUtil.DoTryFilterCatch(TryDelegate tryDelegate, FilterDelegate filterDelegate, CatchDelegate catchDelegate)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.CatchAndFireOnError(UserCodeDelegate function) during the call with ID "Y2M5Y2ZhYzYzMzc4ZTVhYmE0ZjE3MmVjYTVmYmRiZTM.". This exception occurred at the Microsoft Exchange Speech Engine VoIP
platform during an event-based asynchronous operation submitted by the server. The server will attempt to recover from this exception. If this warning occurs frequently, contact Microsoft Product Support.
Anyone with ideas that can help me forward?
//Ase -
Multiple Vertical Axis Renderers and auto scaling
Hi
I have a problem which has been kicking my rear end for a while now.
I have a Linechart to which I add (programmatically in AS code) multiple PlotSeries. I let the first series be handled by the default verticalaxisrenderer (on the left side of the graph) and then the next series (if a different type of data) be rendererd by another verticalAxisrenderer (created programmtically) and added to the right side. And it "works", but the problem is this:
Say I add a series that has a range of 30-80 first. It works fine, BUT when i add a second series with a range of say 1100-1200 the second series is added fine but the first series is 'rescaled' on the axis to go from 30-80 to 0-1200! Which is obviously wrong! I have made other attempts to fix this and it always rescales the axis (sometimes rescaling the data, sometimes not) I have double checked that baseatZero is false and it is--its just the act of adding that second series which causes this redraw and rescale. What am I missing here? Atatched is code snippet and screenshots of behavior.
On Flex 3.5, btw
public function addSeriesToChart(acLocal:ArrayCollection, constellation:String, station:String, param:String):void
var currentSeries:Array = lcGraph1.series;
var stati:int = util.getStationName(station,currentStations);
var stat:String = currentStationsLong[stati];
var s:Object;
var ls:PlotSeries = new PlotSeries();
ls.displayName = y1_axis + " @ "+stat;
ls.yField = "y1";
ls.xField = "x1";
ls.dataProvider = acLocal;
seriesmin(acLocal);
var stroke:Stroke = util.getLineStroke2(nSeries);
ls.setStyle("stroke", stroke);
ls.setStyle("fill",stroke.color);
ls.setStyle("radius", 2);
ls.setStyle("itemRenderer",new ClassFactory(CircleItemRenderer));
s = ls;
var renderers:Array=lcGraph1.verticalAxisRenderers;
Application.application.export.enabled = true;
//two cases: first series and 2+ series: must handle the default verticalAxisRenderer bug
if (nSeries > 0)
//if second series == first series, dont add a new renderer
if (param != currentGraphParam(acParam1))
var la2:LinearAxis = new LinearAxis();
la2.displayName = y1_axis+" ("+util.getUnits(util.EnglishNametoCBIBSParam(param))+")";
la2.title = y1_axis+" ("+util.getUnits(util.EnglishNametoCBIBSParam(param))+")";
la2.baseAtZero = false;
// la2.maximum = seriesmax(acLocal)+1;
// la2.minimum = seriesmin(acLocal)-1;
// lcGraph1.verticalAxisRenderer[0].axis.maximum = cMax;
// lcGraph1.verticalAxisRenderer[0].axis.minimum= cMin;
ar2=new AxisRenderer();
ar2.axis=la2;
if ((nSeries % 2) == 1)
ar2.placement="right"
else
ar2.placement="left";
renderers.push(ar2);
s.verticalAxis = la2;
lcGraph1.verticalAxisRenderers=renderers;
currentSeries.push(s);
lcGraph1.verticalAxisRenderers=renderers;
lcGraph1.invalidateSeriesStyles();
lcGraph1.series = currentSeries;
else
//create new axis
var la2:LinearAxis = new LinearAxis();
la2.displayName = y1_axis+" ("+util.getUnits(util.EnglishNametoCBIBSParam(param))+")";
la2.title = y1_axis+" ("+util.getUnits(util.EnglishNametoCBIBSParam(param))+")";
la2.baseAtZero = false;
ar2=new AxisRenderer();
//dont show it
ar2.visible = false;
ar2.placement = "left";
la2.maximum = seriesmax(acLocal)+1;
la2.minimum = seriesmin(acLocal)-1;
cMax = la2.maximum;
cMin = la2.minimum;
// Alert.show(cMax);
// Alert.show(cMin);
ar2.axis=la2;
renderers.push(ar2);
s.verticalAxis = la2;
currentSeries.push(s);
lcGraph1.series = currentSeries;
lcGraph1.verticalAxisRenderers=renderers;
lcGraph1.invalidateSeriesStyles();
lcGraph1.verticalAxis.title = y1_axis+" ("+util.getUnits(util.EnglishNametoCBIBSParam(param))+")";
disableDefaultVerticalRenderer();
Application.application.arrAc.push(acLocal);
nSeries++;Adobe Flex LiveDocs seemed to indicate "Using multiple series in the same chart works best when the data points are in a similar range (such as a stock price and its moving average). When the data points are in numerically very different ranges, the chart can be difficult to understand because the data is shown on a single axis. The solution to this problem is to use multiple axes, each with its own range. You can plot each data series on its own axis within the same chart using the techniques described in Using multiple axes" and that link is here:
http://livedocs.adobe.com/flex/3/html/help.html?content=charts_types_11.html
I was going to tae a look at this myself, but the code posted here is quite complex, and I suspect incomplete.
If you refer to that link and still cannot solve the issue, I would try your best to boil down your code to a more simple example still exhibiting the issue, and then post that, along with any data and the simplified main app.
If this post answers your question or helps, please mark it as such.
Greg Lafrance - Flex 2 and 3 ACE certified
www.ChikaraDev.com
Flex / AIR Development, Training, and Support Services -
All off-silte call directly goes to Auto Attendant
Hello everyone,
I have an issue with UC520. There is one PSTN line connected to the voice port 0/2/0, All dial out works fine, All off-site calls goes directley to the Auto Attendant, however, interal dial-in works fine, I mean user can dial internal extension properly but not from offsite to insite.
I was wondering if any one can help me.
Here is the partal UC configuration:
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.01.13 13:51:51 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 31685 bytes
dot11 syslog
dot11 ssid uc520-data
vlan 1
authentication open
dot11 ssid uc520-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.1.1 10.1.1.10
ip dhcp excluded-address 192.168.10.1 192.168.10.10
ip dhcp pool phone
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
ip dhcp pool data
import all
network 192.168.10.0 255.255.255.0
default-router 192.168.10.1
ip name-server 63.203.35.55
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp feature access-code
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_BRI
translation-profile outgoing PROFILE_ALL_BRI
voice call send-alert
voice rtp send-recv
voice service voip
sip
no update-callerid
voice class codec 1
codec preference 2 g729r8
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice register global
max-dn 56
max-pool 14
voice translation-rule 4
rule 15 /^...$/ /0354434848/
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^0/ /*/
voice translation-rule 2222
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_BRI
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC520W-8U-2BRI-K9 sn FHK131827A2
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$TC0B$LXMORw4u1vQpD/2eJdN4W1
username admin privilege 15 password 0 admin
username parham privilege 15 password 0 parham
ip tftp source-interface Loopback0
translation-rule 22
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
ip address dhcp
ip access-group 104 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/8
switchport mode trunk
no ip address
macro description cisco-switch
interface BRI0/1/0
no ip address
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
interface BRI0/1/1
no ip address
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
interface Dot11Radio0/5/0
no ip address
ssid uc520-data
ssid uc520-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
ip address 192.168.10.1 255.255.255.0
ip access-group 102 in
ip nat inside
ip virtual-reassembly in
interface BVI100
description $FW_INSIDE$
ip address 10.1.1.1 255.255.255.0
ip access-group 103 in
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone AU
voice-port 0/0/1
cptone AU
voice-port 0/0/2
cptone AU
voice-port 0/0/3
cptone AU
voice-port 0/1/0
cptone AU
voice-port 0/1/1
cptone AU
voice-port 0/2/0
translate calling 1112
connection plar opx 398
description Configured by CCA 4 FXO-0/2/0-Custom-AA
caller-id enable
voice-port 0/2/1
connection plar opx 398
description Configured by CCA 4 FXO-0/2/1-Custom-AA
caller-id enable
voice-port 0/2/2
connection plar opx 398
description Configured by CCA 4 FXO-0/2/2-Custom-AA
caller-id enable
voice-port 0/2/3
connection plar opx 398
description Configured by CCA 4 FXO-0/2/3-Custom-AA
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec711
codec g711alaw
codec g711ulaw
maximum conference-participants 32
maximum sessions 2
associate application SCCP
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 300
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 398
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 2012 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 739
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 90 pots
description AU-Mobile
preference 1
destination-pattern 04........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 68 pots
description NSW Number
preference 1
destination-pattern 02........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 69 pots
description TAS Number
preference 1
destination-pattern 03........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 70 pots
description WA-SA-NT number
preference 1
destination-pattern 08........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 72 pots
description QA-number
preference 1
destination-pattern 07........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 74 pots
description International number
preference 1
destination-pattern 0011T
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 30 pots
description Australia-1800
preference 1
destination-pattern 1800......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 31 pots
description Australia-1300
preference 1
destination-pattern 1300......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 32 pots
description 13 Australia
preference 5
destination-pattern 13....
port 0/2/0
forward-digits all
dial-peer voice 67 pots
description mel-number
preference 1
destination-pattern 9.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 75 pots
description mel-Number
preference 1
destination-pattern 8.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 76 pots
description VIC number
preference 1
destination-pattern 5.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 33 pots
description Emergency NUmber
preference 1
destination-pattern 0000
port 0/2/0
forward-digits all
no sip-register
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
max-ephones 14
max-dn 56
ip source-address 10.1.1.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 7
system message UC520
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/CCMCIP/authenticate.asp
load 7906 SCCP11.9-2-1S
load 7911 SCCP11.9-2-1S
load 7931 SCCP31.9-1-1SR1S
load 7960-7940 P00308010200
load 521G-524G cp524g-8-1-17
time-zone 48
date-format dd-mm-yy
voicemail 300
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$NPt8$6I2moMN32fQoz083VCFm90
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 0.T
transfer-pattern .T
secondary-dialtone 0
night-service day Sun 17:00 09:00
night-service day Mon 17:00 09:00
night-service day Tue 17:00 09:00
night-service day Wed 17:00 09:00
night-service day Thu 17:00 09:00
night-service day Fri 17:00 09:00
night-service day Sat 17:00 09:00
create cnf-files version-stamp 7960 Dec 23 2013 10:55:20
ephone-template 15
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
button-layout 7931 2
ephone-template 16
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
ephone-template 17
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
ephone-template 18
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
button-layout 7931 2
ephone-dn 5 dual-line
number 301 no-reg primary
label 301
description PhoneA Analog
name PhoneA Analog
ephone-dn 6 dual-line
number 302 no-reg primary
label 302
description PhoneB Analog
name PhoneB Analog
ephone-dn 7 dual-line
number 303 no-reg primary
label 303
description PhoneC Analog
name PhoneC Analog
ephone-dn 8 dual-line
number 304 no-reg primary
label 304
description PhoneD Analog
name PhoneD Analog
ephone-dn 9
number BCD no-reg primary
description MoH
moh out-call ABC
ephone-dn 10 dual-line
number 201 no-reg primary
pickup-group 1
label 201
description Extension 201
name Receptionist Receptionist
mobility
call-forward busy 300
call-forward noan 300 timeout 20
ephone-dn 11 dual-line
number 207 no-reg primary
label 207
description Extension 207
name None None
ephone-dn 12 dual-line
call-waiting ring
number 203 no-reg primary
pickup-group 1
label 203
description Extension 203
name Peter Steve
call-forward busy 300
call-forward noan 300 timeout 15
huntstop channel
ephone-dn 13 dual-line
call-waiting ring
number 204 no-reg primary
pickup-group 1
label 204
description Extension 204
name Tim OConnor
call-forward busy 300
call-forward noan 300 timeout 20
huntstop channel
ephone-dn 14 dual-line
number 205 no-reg primary
pickup-group 1
label 205
description 205
name 205
ephone-dn 15 dual-line
number 206 no-reg primary
pickup-group 1
label 206
description 206
name 206
ephone-dn 16 dual-line
call-waiting ring
number 202 no-reg primary
pickup-group 1
label 202
description Extension 202
name David Holmes
call-forward busy 300
call-forward noan 300 timeout 15
huntstop channel
ephone-dn 17 dual-line
number 208 no-reg primary
label 208
description 208
name 208
ephone-dn 18 dual-line
number 209 no-reg primary
label 209
description 209
name 209
ephone-dn 19 dual-line
number 210 no-reg primary
label 210
description 210
name 210
ephone-dn 43 octo-line
number 771 no-reg primary
conference meetme
preference 3
ephone-dn 44 octo-line
number 771 no-reg primary
conference meetme
preference 2
no huntstop
ephone-dn 45 octo-line
number 771 no-reg primary
conference meetme
preference 1
no huntstop
ephone-dn 46 octo-line
number 771 no-reg primary
conference meetme
no huntstop
ephone-dn 49 octo-line
number C001 no-reg primary
conference ad-hoc
preference 3
ephone-dn 50 octo-line
number C001 no-reg primary
conference ad-hoc
preference 2
no huntstop
ephone-dn 51 octo-line
number C001 no-reg primary
conference ad-hoc
preference 1
no huntstop
ephone-dn 52 octo-line
number C001 no-reg primary
conference ad-hoc
no huntstop
ephone-dn 55
number A801... no-reg primary
mwi off
ephone-dn 56
number A800... no-reg primary
mwi on
ephone 1
device-security-mode none
mac-address 4142.4DB8.0000
ephone-template 16
max-calls-per-button 2
type anl
button 1:5
ephone 2
device-security-mode none
mac-address 4142.4DB8.0001
ephone-template 16
max-calls-per-button 2
type anl
button 1:6
ephone 3
device-security-mode none
mac-address 4142.4DB8.0002
ephone-template 16
max-calls-per-button 2
type anl
button 1:7
ephone 4
device-security-mode none
mac-address 4142.4DB8.0003
ephone-template 16
max-calls-per-button 2
type anl
button 1:8
ephone 5
device-security-mode none
mac-address 0024.97AA.E811
ephone-template 15
max-calls-per-button 2
username "Receptionist" password receptionist
type 7931
button 1:10
--More-- !
ephone 6
device-security-mode none
mac-address 0024.C4FC.4013
ephone-template 16
username "None"
type 7911
button 1:11
ephone 7
device-security-mode none
video
mac-address 000F.34FA.168B
ephone-template 16
username "steve" password petersteve
speed-dial 1 xxx label "Peter - Home"
speed-dial 2 xxx label "David - Mobile"
speed-dial 3 xxx label "Tim - Mobile AUS"
speed-dial 4 xxx label "Tim - Mobile USA"
type 7960
button 1:12
ephone 8
device-security-mode none
video
mac-address A40C.C394.B1F0
ephone-template 16
username "tim" password timoconnor
speed-dial 1 xxx label "David - Mobile"
speed-dial 2 xxx label "Peter - Mobile"
speed-dial 3 xxx label "Clare - Mobile"
type 7911
button 1:13
ephone 9
device-security-mode none
mac-address 0024.C4FC.5425
ephone-template 16
type 7911
button 1:14
ephone 10
device-security-mode none
mac-address 0024.C4FD.E27C
ephone-template 16
type 7911
button 1:15
ephone 11
device-security-mode none
video
mac-address 0007.5098.1AB6
ephone-template 16
username "holmes" password davidholmes
speed-dial 1 xx label "David - Home"
speed-dial 2 xxxl abel "Sue - Mobile"
speed-dial 3 xxx label "Peter - Mobile"
speed-dial 4 xxx label "Tim - Mobile USA"
speed-dial 5 xxx label "Tim - Mobile AUS"
type 7960
button 1:16
ephone-hunt 1 sequential
pilot 501
list 202, 203, 204
final 300
timeout 8, 8, 8
no-reg pilot
statistics collect
description Sales
alias exec cca_vm_notification schedule from_time=00 to_time=24
banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Nov 15 22:54:23 EST 2013^C
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
transport input all
line vty 5 100
transport input all
ntp master
end
UC520#I was configure custom disconnect tone refer to this site:
http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html
And the tone is in the attachment, and the custom disconnect tone configuration like below:
voice class custom-cptone Disconnect
dualtone disconnect
frequency 420 420
cadence 251 255 245 250 249 250 250 250
and the port configuration was like below:
voice-port 0/1/2
supervisory disconnect dualtone mid-call
supervisory custom-cptone Disconnect
cptone NL
timeouts interdigit 4
timeouts call-disconnect 5
timeouts wait-release 5
timing hookflash-out 500
connection plar 334
impedance complex2
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
but it was not working and the phone still ringing after the PSTN caller disconnect.
but i was read about "dualtone-detect-params", and i was add the below command and i do not understand it, but it was solve hte problem:
voice class dualtone-detect-params 1
freq-max-deviation 20
cadence-variation 50
so what it is and how to determine this parameters. -
Using Skype as a auto attendant / switchboard
Is there a way to use a Skype number to call multiple skype users, so the first one to pick up takes the call? Also is there any auto-attendant capability - so that you can have some press 1 for one extension and 2 for another etc.
This needs to be made available. Auto attendant is a good feature for skype and its a good option for companies to use if they have high call volumes.
We wish this would be done within 1 year(s)
Thanks!
Regards,
Jacob's Sales & Tech Store -
Turn Auto Attendant into Night Mode Manually
I have set an auto attendant to automatically play the day and night mode based on business hours on Unity Connection with CUCM 8.0. But client wants to turn on the night mode manually when the receptionist is leaving before the closing hour, and turn off the night when come in early. And they still want to play the day greeting during normal business hours. Which option available in Unity Connection that can handle this. I can set up night mode manually ok, but will lose out the day greeting. How can I achieve this with both day and night greeting.
Thanks.
Dat PhamConfigure a separate call handler and have the recepcionist set CFA for this.
All she needs to turn this off/on is to set CFA to a predefined DN.
CUC can only work based on pre-defined schedules.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk -
Auto attendant intermittently routes call to out of region/not in dial plan UM server
Hi all,
Exchange 2013 on prem, hardware not virtual. CU5 w/Lync 2013
I've got calls that get intermittently routed to UM servers that are out of region and not in the dial plan. The out of region UM server sees the call is outside of business hours & sends helpdesk calls to voicemail instead of the appropriate phone
menu.
Additionally, when Exchange admins who are in different time zones look at the GUI w/the AA's business hours they see a time skew even though the time displayed is listed as Eastern. I think the mis-routing & the time zone skew are related. When
the Tokyo server gets the call it checks the time: 3AM? Not in business hours even though in Eastern Time where the call is supposed to go, it is in business hours.
In the Lync client log (as seen via the snooper tool) this is the last message before the call gets transferred:
“ms-diagnostics:
15032;reason="Re-directing request to the destination in 302”
Additionally the time zone on the AA schedule is set to Eastern Time. Why is the TYO UM server ignoring this and applying local time?
Any tips to point me in the right direction would be appreciated.
AdamNumber two was correct! The affected site did not have an arbitration mailbox. Details follow.
I still have the underlying problem of AA's getting the time zone of the UM server applied rather than the time zone they are allegedly set to (for example Beijing business hours served from a TYO UM server getting TYO time).
With the help of MS support we resolved the immediate problem: calls getting routed to our TYO site.
It turns out that every AD site with Exchange servers needs to have an arbitration mailbox with the grammar generator role set & ready. If a site with UM servers does not have an arbitration mailbox it will proxy the call to another site that does.
In our case, it would route them to our Tokyo site that applied the wrong hours to the auto attendant.
Here's how we created the arbitration mailbox
[PS] C:\temp\autoattendant>New-Mailbox -Arbitration -Name "A new UM Grammar Mailbox" -Database <some db hosted in site> -UserPrincip
alName [email protected] -DisplayName "A new UM Grammar Mailbox"
C:\temp\autoattendant>Set-Mailbox [email protected] -Arbitration -UMGrammar:$true
This keeps the call from going out of site to an Exchange UM server in a different time zone.
The tricky bit is that this does not immediately work. The mailbox needs to pick up the OrganizationCapabilityUMGrammarReady capability which it will only get when the grammar generator runs. In 2010 you were able to kick this off manually. In
2013 it runs once a day. You have to wait until Get-Mailbox -Arbitration | fl name, servername, persistedcapabilities shows the OrganizationCapabilityUMGrammarReady has been assigned to the mailbox.
I still have not yet resolved the underlying problem of why UM servers are ignoring the time zone setting on AA's business hours. -
Hello,
When an incoming call hits the UC, I want to switch manually (button or code) what it does.
Normally during day the calls go to the Auto Attendant.
During night there is a message.
During the break there is another message.
And they want to have also 2 other messages they want to use in other cases.
Is it possible to resolve this problem with a script of some kind?
If not I need to let the call come in on a phone and set the phone in call-forward to the AA or different voicemail boxes through floating extensions.
Thanks,
StephanHello,
This is going to be a little tricky. There is a couple of ways this can be accomplished, but there are some limitations with each method.
You can use the opened/closed hours in the Auto Attendant to change the greetings. The limitation here is that you can't manually switch, only be a schedule.
The alternative, is to use a manual night service configuration to send calls to 2 separate auto attendants. This will allow manual control, but only between 2 messages.
This might be possible with a custom script, but as we don't support custom scripts with UC500s, I couldn't tell if that would work.
Hope this helps.
Thanks,
-john -
Unity Connection Auto Attendant direct to voice mail, MWI issue
hi,
First time encountering this issue with call transfer from AA and MWI.
An Auto Attendant have been setup for a branch office. All employees at the Branch office phones are assigned a DID.
Customers can dial the 10 digit or dial the main line (auto attendant) and then dial the last 4 digit extension to reach the employee.
Problem I am having is there is a range of numbers that when being transferred from AA, goes directly to voicemail. Also when there is a message left on the voicemail, the MWI is not lit. It does show the envelope icon and the message "you have vociemail" on the phone display.
I can call the DID directly and it does not go directly to voicemail but MWI is still not on when a message is left.
This only happens within a range of numbers.. for example, I have 50 DID assigned to this office from XXX-XXX-0100 to XXX-XXX-0150. Only the phones within the range of XXX-XXX-0120 to XXX-XXX-0130 have this issue with the AA going directly to vociemail and MWI not on when a message is left.
For phones outside this range of numbers, AA transfer works fine and MWI works.
anyone ran into this issue before?
thank in advance.Hi,
The range in which you are facing issue is XXX-XXX-0120 to XXX-XXX-0130, just try to open any one of the extension in this range and go to edit -- transfer rules -- standard/alternate (if alternate is enabled) -- Transfer calls -- may be this is set to Greetings. If so the transferred call goes to voicemail directly. Try enabling Extension.
Just try this for one extension and see...
Hope this helps, i had the same issue.
Regards
Jagadish G -
Call forward to external number which has auto attendant
Hi
I am a voice administrator in my company
I want to forward all of my calls to my Other location's number.
Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu.
Is it possible to enter a digit pattern in call forward destination in CUCM so that it can take care of Auto attendant menu of my Other location and land on my number?
We have CUCM 8 running.
Please help!!
AshwinHello Ashwin,
"Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu."
What type of phone system and voice mail is providing the auto attendant?
How are the POTS lines(analog only correct?) terminated into the phone system?
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