Retrieve UM Auto Attendant configuration
in
this-link there are many shell commands for editing UM Auto Attendant configuration ,
but there are no commands for retrieve the same information ... is EWS API is the answer?
I'm trying with C# / shell commands to get these information
How I :
get UMAutoAttendant Business Hours ?
get Business Hours Greeting ?
etc...
for example you can change business hours schedule , but you can't get the same data
Set-UMAutoAttendant -Identity MyUMAutoAttendant -BusinessHoursSchedule 0.10:45-0.13:15,1.09:00-1.17:00,6.09:00-6.16:30
I wanna retreive the same information
0.10:45-0.13:15,1.09:00-1.17:00,6.09:00-6.16:30
the only thing i can get is
[PS] C:\Windows\system32>Get-UMAutoAttendant -Identity ZakosUM
Name UMDialPlan PilotIdentifierList SpeechEnabled Status
ZakosUM DP {} True Enabled
thanks
You can execute the cmdlets from Managed code using Remote powershell
http://msdn.microsoft.com/en-us/library/office/ff326159(v=exchg.140).aspx
eg
System.Security.SecureString secureString = new System.Security.SecureString();
foreach (char c in Password)
secureString.AppendChar(c);
PSCredential credential = new PSCredential(AdminUserName, secureString);
WSManConnectionInfo connectionInfo = new WSManConnectionInfo(new Uri("https://" + PSServerName), "http://schemas.microsoft.com/powershell/Microsoft.Exchange", credential);
connectionInfo.AuthenticationMechanism = AuthenticationMechanism.Basic;
connectionInfo.SkipCACheck = true;
connectionInfo.SkipCNCheck = true;
connectionInfo.MaximumConnectionRedirectionCount = 4;
Runspace runspace = System.Management.Automation.Runspaces.RunspaceFactory.CreateRunspace(connectionInfo);
runspace.Open();
Pipeline plPileLine = runspace.CreatePipeline();
Command gmUMAutoAttendant = new Command("Get-UMAutoAttendant");
gmUMAutoAttendant.Parameters.Add("Identity", "test");
plPileLine.Commands.Add(gmUMAutoAttendant);
Collection<PSObject> RsResultsresults = plPileLine.Invoke();
foreach (PSObject obj in RsResultsresults)
PSObject bhs = (PSObject)obj.Properties["BusinessHoursSchedule"].Value;
ArrayList bhsObj = (ArrayList)bhs.BaseObject;
foreach(String Element in bhsObj){
Console.WriteLine(Element);
plPileLine.Stop();
Cheers
Glen
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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. -
Full Directory option on auto attendant consoles
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I have already configured the settings on server to sync contacts.
Please advise.
ThanksYes, all End Users (not Application Users) from CUCM should appear in the CUxAC directory provided that it meets any rules you have configured. By default it should pull everything over. This does not happen in realtime so you may need to force a sync, most commonly through a service restart.
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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 -
Auto-Attendant Not Running in UC500 Series
Hi guys,
I've got a little problem here. My auto-attendant couldn't run in my UC500.
Here I give you a little command I took from my UC500:
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern 57973199$
b2bua
voice-class sip outbound-proxy ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
Please let me know if you need any more command from my UC.
Regards,
Dennyupdated,
There is no information in service-module integrated-Service-Engine 0/0 status.
I think this is the part of voicemail configuration (CMIIW)
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 221
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
If I telnet the UC via CME, the user interface looks so simple, very2 different with the old one.
Actually, I don't care with the voicemail, but I just care with the auto attendant. Because the ip phone cannot be dialled from outside using its direct line, but can be dialled from inside.
I really appreciate if you can give me another advice for this one.
Regards,
Denny -
UC520 FXO To Auto Attendant Disconnect Problem
We have UC520 with FXO card, and have a problem when we point the inbound call to the Cisco Unity Express Auto Attendant, the problem is if the PSTN caller disconnect, the FXO remain off-hook for along time, and if the PSTN caller dial an extension and then disconnect the phone remain ringing for a long time.
but if we point the inbound call an extention, there is no problem and the phone stop ring when the PSTN caller disconnect.
below is the voice-port, and CUE dial-peer configuration:
voice-port 0/1/2
translate calling 3
compand-type a-law
cptone NL
timeouts interdigit 4
timeouts call-disconnect 5
timeouts wait-release 5
timing hookflash-out 500
connection plar 489
impedance complex2
description LandLine 5105586
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
voice-port 0/1/3
translate calling 3
supervisory disconnect dualtone mid-call
compand-type a-law
cptone NL
timeouts interdigit 4
connection plar 489
impedance complex2
description LandLine 5105388
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
dial-peer voice 499 voip
description VoiceMail
destination-pattern 498
media flow-around
session protocol sipv2
session target ipv4:10.1.10.2
dtmf-relay sip-notify
codec g711ulaw
no vad
so what we can do?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. -
Removal of Auto Attendant ....
Hi Gurus,
I am having some problems with the UC 520 namely,
1) Is it possible to remove the auto attendant?
2) I would like when people call in to the main line, it would be diverted to a voice hunt group.
The voice hunt group has already been configured as show below,
voice hunt-group 1 sequential
final 3399
list 3309,3317,3318,3315
timeout 10
pilot 3380I am not sure what do you mean by E1/T1?
Indeed i am using a DID service.
Here you go,
dial-peer cor custom
name internal
name local
name domestic
name international
name 900
name 976
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-domestic
member domestic
dial-peer cor list call-international
member international
dial-peer cor list call-900
member 900
dial-peer cor list call-976
member 976
dial-peer cor list user-internal
member internal
dial-peer cor list user-local
member internal
member local
dial-peer cor list user-domestic
member internal
member local
member domestic
dial-peer cor list user-international
member internal
member local
member domestic
member international
dial-peer cor list user900-internal
member internal
member 900
member 976
dial-peer cor list user900-local
member internal
member local
member 900
member 976
dial-peer cor list user900-domestic
member internal
member local
member domestic
member 900
member 976
dial-peer cor list user900-international
member internal
member local
member domestic
member international
member 900
member 976
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 3398
b2bua
session protocol sipv2
session target ipv4:IP ADDRESS
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 150 pots
description ** incoming dial peer **
translation-profile incoming INCOMING_DID
incoming called-number .%
direct-inward-dial
port 0/3/0:15
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 3399
b2bua
session protocol sipv2
session target ipv4:IP ADDRESS
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 400 voip
description Office Tele
destination-pattern 4..
voice-class codec 1
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 500 voip
description TeleInformatics Voicemail
destination-pattern 5..
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 6 voip
description Oilexec
preference 7
destination-pattern 6..
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 31 voip
description London Office internal Extensions 31**
destination-pattern 31..
voice-class codec 1
session protocol sipv2
session target ipv4:IP ADDRESS
dtmf-relay sip-notify
dial-peer voice 32 voip
description Jersey Office internal Extensions 32**
destination-pattern 32..
voice-class codec 1
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 199 voip
description Aberdeen Reception phone
destination-pattern 199
voice-class codec 2
session protocol sipv2
session target ipv4:IP ADDRESS
dtmf-relay sip-notify
dial-peer voice 151 pots
corlist outgoing call-local
description ** E1 pots dial-peer **
translation-profile outgoing OUTGOING_DID
preference 5
destination-pattern 9[5-9]T
port 0/3/0:15
no sip-register
dial-peer voice 153 pots
corlist outgoing call-international
description ** E1 pots dial-peer **
translation-profile outgoing OUTGOING_DID
preference 5
destination-pattern 9001T
port 0/3/0:15
prefix 001
no sip-register
dial-peer voice 8 voip
description SIP Link to Regent Quay - ABZ Dial-tone
translation-profile outgoing VOIP_ABERDEEN
destination-pattern 8T
voice-class codec 1
session protocol sipv2
session target ipv4:IP ADDRESS
dtmf-relay sip-notify
clid network-number 12345678912
dial-peer voice 201 voip
description C-MAR internal Extensions 2**
destination-pattern 2..
voice-class codec 2
session protocol sipv2
session target ipv4:172.16.0.1
dtmf-relay sip-notify
dial-peer voice 154 pots
corlist outgoing call-international
description ** E1 pots dial-peer **
translation-profile outgoing OUTGOING_DID
destination-pattern 9019T
port 0/3/0:15
prefix 019
no sip-register
dial-peer voice 155 pots
corlist outgoing call-local
description ** E1 pots dial-peer **
translation-profile outgoing OUTGOING_DID
destination-pattern 91800T
port 0/3/0:15
prefix 1800
no sip-register
dial-peer voice 156 pots
corlist outgoing call-local
description ** E1 pots dial-peer **
translation-profile outgoing OUTGOING_DID
preference 5
destination-pattern 9100
port 0/3/0:15
forward-digits 3
no sip-register
dial-peer voice 2002 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 3397
b2bua
session protocol sipv2
session target ipv4:10.1.12.1
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 2003 pots
destination-pattern 3301
port 0/0/0
dial-peer voice 10 pots
destination-pattern 3301
port 0/0/0
dial-peer voice 157 pots
description ** FAX 1**
destination-pattern 9012T
port 0/3/0:15
prefix 012
dial-peer voice 158 pots
description ** FAX 2 **
destination-pattern 9013T
port 0/3/0:15
prefix 013
dial-peer voice 159 pots
destination-pattern 916..
port 0/3/0:15
forward-digits 4
dial-peer voice 34 voip
description wellington Office internal Extensions 34**
destination-pattern 34..
voice-class codec 1
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 35 voip
description Dubai Office internal Extensions 35**
destination-pattern 35..
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 37 voip
description Mumbai Office internal Extensions 37**
destination-pattern 37..
voice-class codec 1
session target ipv4:IP ADDRESS
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 30 voip
description Houston Office internal Extensions 30**
destination-pattern 30..
voice-class codec 1
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
no vad
dial-peer voice 38 voip
description Brasil Office internal Extensions 38**
destination-pattern 38..
session target ipv4:IP ADDRESS
dtmf-relay cisco-rtp
codec g711ulaw
no vad
no dial-peer outbound status-check pots
sip-ua
registrar ipv4:172.16.3.1 expires 3600 -
Hi Guys,
I am configuring Exchange Auto Attended (Exchange 2013) and I've managed to get all of my personalized prompts in place but there is one exchange default one that I can't seem to find to disable...It's the one that says
"I couldn't hear you, please say the name of the person you want to contact, or the option you'd like to choose"
That final prompt doesn't sound professional, I am wondering if it's part of some kind of timeout or if it can even be changed.
THanks
BryanHow have you been disabling them? You can search for the prompt in the <Program Files>\Microsoft\Exchange Server\V15\UnifiedMessaging\Prompts\<language>. They're pretty well named, once I find it, I'll usually
rename it .old and make a copy of the smallest blank one that's in the directory (sort by size) with the same name as the one you want to replace. Of course, you'll have to rinse and repeat on potentially many servers here and watch that it doesn't come
back with a patch, but it's one method.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications -
Auto-attendant on CUCM 7.0
Hello,
is there anyway to configure autoattendant on CUCM 7.0?
ThanksHi javalenc,
I have implemented CUCMBE 7.1 on a linux system with unity.
Everythink works perfect.
But I need to configure auto attendant that can:
Answer a call
Play a user-configurable welcome prompt
Plays a main menu prompt that asks the caller to perform one of three actions:
Press 0 for the operator.
Press 1 to enter an extension number.
Press 2 to spell by name.
Please I need all the steps to through up to end. If possible the screen show of the steps. -
If anyone has some constructive words of advice on this issue, I would greatly appreciate them. I have come to a road block on this one issue with my SPA9000. It seems that after changing my scripts and messages my Auto Attendant is no longer available, it just gives a busy signal. So inbound callers get a busy signal, since its set to go to "aa" on inbound calls. I've done lots of things to try to pin point the issue including rolling back to a previous configuration(scripts etc.), and the only consistent thing I've noticed is both Auto Attendant( Directory -> Corporate Directory -> Auto Attendant) and pbx2 are responding with a busy signal. Even after a Factory Reset. When I was setting up my AA messages I did make a mistake and dialed "*72" instead of "****" then 72...(on FXS 2). That is the only possible cause from a human error perspective, that may have caused this issue. I've tried to see what that combination of key strokes may have done but there is no documentation (in my possession) that gives any insight. So if anyone may know where to look or has run into this before could you please point me in the right direction. -Thanks
not sure if this will help but when i ran into an AA problem before this resolved it...not sure why, but here it is anyway...
under the AA parameters of SIP tab enter a value on the Daytime field (For example, start=9:0:0;end=17:0:0 means the start time is 9 AM and the end time is 5 PM. The other hours (5 PM to 9 AM) are considered nighttime hours.)
DayTime AA: Yes (default)
DayTime AA Script: 1 (default)
Also, check if Current AA under Info tab (Auto Attendant Prompt Status ) states Daytime... -
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:
Submission to the Hub Transport server failed. The operation will be retried. ---> Microsoft.Exchange.Net.ExSmtpClient.UnexpectedSmtpServerResponseException: Unexpected SMTP server response. Expected: 220, actual: 500, whole response: 500 5.3.3 Unrecognized
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 -
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 -
Ring Duration Before Auto Attendant
Hi,
I have Call Manager Express with Integrated Unity 8.0 and i wants to configure ring duration before auto answer the call. How to configure this please?I dont need any ringing i need direct to auto answer whenever calling from outside from auto answer i will select the extension like below message. I need the command to configure in cme or integrated cue.
http://docs.fortinet.com/fvox/cli-html/2-2-0/index.html#page/FortiVoice%20Online%20CLI%20Reference/config.3.39.html
ring-duration
Enter the number of seconds for the phone to ring before the auto attendant answers with the greeting message.
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