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.

Similar Messages

Ringing duration before call forwarding

I have a skype number with a paid subscription. I use a skype phone on my home network to receive business calls on. In the previous skype UI I set the ringing duration to 25 seconds before the call would forward to my mobile number. It works really well for me when I'm out and about. However I want to change the ringing duration before diversion to 15 seconds. On the new Skype UI this option has disappeared (or at least I can't find it anywhere!?) Any help would be much appreciated. Thanks.

Hi, k1092, and welcome to the Community,
Remember, the setting is not for the number of rings before the incoming fall is forwarded, but the amount of time. In the Windows version of the Skype software, Tools -> Options -> Calls -> Call Forwarding -> Call Forwarding Options. Remember to Save any changes you make to your settings.
You may first need to check the timing on how long the number of times the incoming call chime sounds (rings).
Best regards,
Elaine
Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

UC320 Auto Attendant answering delayed calls to fast.

I recently deployed a UC320W with the latest firmware upgrade (2.3.2). My trouble is the auto-attendant intermittently answers delayed ringing calls on the first or second ring no matter what the timer is set to (which is 25 sec.). Doesn't seem to be any particular line. There are 4 FXOs on the UC320 and a 5th FXO on a SPA8800. They are all configured as shared FXOs in key system fassion. I've done impedance matching on all 5 FXO lines.
This has happened twice before and I was able to workaround by downgrading firmware to 2.2.2.
I can't do that in this case, however, because a few of my phones are SPA512Gs, which, it's my understanding, only work with the 2.3.2 firmware.
Right now, I simply have the delayed ringing disabled.
We have another technician in another state that tells me he has run into the same issue as well.
Any suggestions?

Hi Carlos,
Please clarify the issue you are having.
My understanding is that you have 5 FXO lines configured as shared FXO lines and call is forwarded if no answer to AA, timer is set to 25 seconds. AA answers the call intermittently on the first or second ringing before 25 seconds timer expries.
Please send me UC320w feedback at Services (on the top of configure utility) -> feedback, put "attention [email protected]" at Issues or Suggestions section. So that I will be able to look into your configuration and understand your issue better.
Best regards,
Wendy

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

AUto attendant not working

Auto attendant not working .....
If i assign AA script Transfer_v02.aef then i hear an just 1-2 seconds the script after that again it is just ringing ...
Is there Option to set
Please press the extension which you want to Call ...
Eg: if i press 222 that time it should ring the 222 extension
Please help this is the first time i am working on this stuff
If i chnage the scrpt file then i can hear auto attendant not initiazlised.. please find the attched screen shot and back file for more details.
Best Regards
Shabeel

Hi KMS, check the codec used between that GW using the FXO and the CTI RP.
Check also the Calling Seach Space
GW should be able to reach CTI Route Point and CTI ports partition.
If you try other extension it works fine?

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.

How do I extend incoming call duration before voicemail activates ?

How do I extend incoming call duration before voicemail activates ?

I had this problem and for 12 months no one could tell me how to fix it.
Finally a young kid told me. Get onto your carrier web site, in this case Telstra in Australia,
and you will search for ringing or similar and there should be instructions how to do it. It worked!!!
good luck

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.

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 Pham

Configure 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

SPA9000 Auto Attendant busy

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...

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.
Adam

Number 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.

SPA9000 Auto Attendant Answering problems

Hi All Have a problem on auto attendant It answers all calls properly (my voice saying please enter the extension you require) but after typing in the extension my voice comes on again saying sorry the person you are calling is not available. This is not true the IP phone rings no problem. How do I stop this I cannot type in the AA script as it says that its invalid HTML Thanks everybody Marky

can you provide additional details like the model number of the IP phone you are using? are you using SPA400 as well? is the call coming from PSTN or ITSP? can the IP phones dial each other locally?
if you are using SPA400 please make sure that you have the latest firmware on both SPA400 and SPA9000 (1.0.1.4 and 5.2.5 respectively)
also, if you have changed the AA script using set-up wizard try the default script which you can find below just in case the wizard has messed with the script...
<aa><form id="dir" type="menu"><audio src="prompt1" bargein="T"/><noinput timeout="10" repeat="T"/><nomatch repeat="F"><audio src="prompt3" bargein="T"/></nomatch><dialplan src="dp1"/><match><default><audio src="prompt2"/><xfer name="ext" target="$input"/></default></match></form></aa>
| isolate! isolate! isolate! |

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!!
Ashwin

Hello 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?

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

E71 - Can't change ring duration

I've been searching for some time for information on extending the length of a ringtone on an E71 beyond the default 5 rings. I've tried numerous setting combinations, ringtones and even created different length custom ringtones but am unable to extend the length. This can be a real problem if I don't have my mobile phone in the same room as me - by the time I get to it chances are the call has already gone through to the message bank.
Is anyone else having this issue?
Does anyone have a solution or workaround?
Thanks
Solved!
Go to Solution.

I am an Orange PAYG customer with an N97 and my phone diverts after 15 seconds, which is a real pain as I miss more calls than I manage to answer.
Having spoken to Orange I was initially given the code to extend, but this didn't work. On contacting them again I am now told that PAYG customers can only have 15 seconds and there is nothing they (or I?) can do about it.
Quite why this restriction applies only to PAYG was not explained, (Contract customers can have 25 seconds) but the money they make from voicemails and call backs is likely to be the major factor.
If I was selecting an operator for the future, Orange would not appear on my list if this is how they treat their customers.
Interestingly, you won't find any reference to ring duration in their FAQ, so they do appear to be ashamed of their policy, but not sufficiently so to take the necessary action.
If there is a fix with third party software, I'd like to know more please.

Ring Duration Before Auto Attendant

Similar Messages

Maybe you are looking for