Auto attendant not forwarding calls to Receiption

Hello,
I am having CME configured in Cisco 2821 router having CUE 3.2 and CME 7.0 installed. When I dial the call from outside the organization,I am able to hear the prompt but when I press 0 to to reach the reception I used to get prompt message as "Your call can not be completed" and then it plays another prompt.
Can anyone help here?

Yes receiptionist phone is located at same site as CUE.
Customer has moved to new location and is a single site but now the PRI got changed, now they are using TTML.
Does configuration has anything related to PRI numbers?
What I need to check in routing?

Similar Messages

Configure Cisco CME auto-attendant to forward call after ext is dialed

I am trying to configure my CME if the auto attendant picks up and a users extension is dialed after hours to forward that call to the number configured for that extension, currently if the extension is dialed the call is not forwarded.
Thank you,
-Tom

Hi,
I have same or similar issue, when a person call my company and press option 3 for example (after hour) the call get forward to a manager mobile number, my question is how can I disable the forwarding from CUCM? the OS i use is
8.6.2.22900-9
Many thanks

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?

HT4994 I have an Iphone 4 and can not forward calls. every time I put the number in the call forward field it jumps back a page and the button moves to off. can anyone assist.

I can not forward calls. I go to call forwarding, turn it on, copy and paste my phone number into the correct field (including the international dialling code).
I then tap the Call Forwarding button, which takes me back one stage and the Call Forwarding button goes to off after a couple of seconds. Anyone who can help?

Yes, a blocked call will go immediately to your voice mail.
If you would rather not have that behavior, ask your cellular carrier if you can block that person through the carrier.

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.

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

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,
Denny

updated,
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

Online Number is not forwarding calls to my mobile

Hi,
I have gone premium and selected unlimited landline for Australia. I have then bought an Online Australian Number. After setting up call forward to my Indian Mobile Number the calls are not getting forwarded, although the calls appear on Skype Client when I'm online. I have tried setting privacy and all that stuff, but it yet does not work. In the Call Forward tab following message is displayed: "All calls forwarded to phones covered by your subscription are included at no extra cost. Other destinations will be charged at our low per minute rates."
Am I missing something? Do I still need to buy credit? Thanks so much,
Kind Regards
Prabh

Hello,
Skype Number enables you to receive calls on your Skype account and the caller will be charged local rates.
To receive calls to your Skype Number, you should be online on your Skype application, or you can enable call forwarding and voice messaging. And also, please make sure that you are signed in to only one device.
Please make sure that you have set up your Privacy Settings to "Received calls from anyone."
Sign in to Skype.
Tap the menu icon .
Select Settings.
Go to Receive calls from option.
If one of my replies has addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others experiencing the same issue.

AutoAttendant not transferring calls in CUE 8.6 to CME 9.1

**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_US

Ok I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
   Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
    IEC Version: 1
    Entity: 1 (Gateway)
    Category: 129 (Call setup timeout)
    Subsystem: 7 (SIP)
    Error: 67 (ACK wait timeout)
    Diagnostic Code: 0
2811-TEST#

Shared DN not forwarding to Call Handler

I'm sure I'm probably over looking something simple but here is my problem.
I have a shared DN that forwards to Unity but instead of the Call Handler greeting I am geeting the Auto Attendant. The Call Handler is set to record a message and send it to a public distribution list.
Anyone out that have any devine insight to my problem?

This is a direct call on and off network to the shared DN.
The routing rule states:
On
Both
Any
Any
2947
Any
Always
Send to greeting for NGAL J6 HELPDESK VOICEMAIL
I do not have the extension listed in the Call Handler.

CUCM not forwarding CTI called number - Subscriber Sign-in

I am running into an issue in CUCM/CUC 8.6(2a) when setting up external voicemail access.
I set up a CTI RP (Dn=2300) to forward to voicemail and set up a routing rule to send the Call
to subscriber sign in. The call keeps going to the opening greeting. I did a port status monitor
and I don't see the forwarding stations directory number. Just the Unity Pilot Point number.
I just set this up in CUCM 8.6(1) and I have it working. I mirrored the config but don't get the same results.
I'm not sure if maybe a service parameter changed or something and I am missing it.
TIA

Thanks for sharing your findings Rob. Your post helped me solve an issue I was having with a SIP trunk between CUCM and CUC.
Just to recap in case this post can help anyone else, if you don't check the Redirecting Diversion Header Delivery - Outbound then Call Manager will not forward calls with a redirect number or reason code to Unity. Using RTMT Port Monitor you'll find the call will complete but as a direct call to Unity.
The problem I was facing was CFNA (internal, external, etc) from a DN in CUCM to CUC would send the caller to the Opening Greeting, instead of the voicemail box of the user, which in this case was functioning properly because Redirecting Diversion Header Delivery was not checked.
Thanks again to both Robs for your thread,
Derek

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?

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.

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?

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.

Custom Auto Attendant Prompts through TUI not working for users who migrated from 2010 to 2013

In Exchange 2010, we started using unified messaging and set up Auto Attendants. We setup a admin role/RBAC for people of a security group to be able to update the message on the auto attendants. They have the UMPrompts assigned role. All of this is working
great in 2010. We have now migrated to 2013, and the users who were migrated from 2010 to 2013 can no longer update the messages through TUI. Newly created 2013 users can and are assigned the EXASCT same permission as the users who have been doing this for
well over a year on 2010. When they call the AA and press #,* they are asked to provide their extension, after doing so the system tells them that extension is not correct. and asks for the extension again. Newly created users with the same permissions
get prompted for their PIN and can log in and change the message just fine.
Confirmed Bug? anybody else having this issue?
What would be different for this process between a user who was migrated from a previous version like 2010 compared to a newer user who has only ever existed on 2013?

What if the migrated 2010 users are in the same DB as the system mailbox? I had a similar issue during a migration; http://www.skypeadmin.com/2014/11/10/known-issue-um-automated-attendant-tui-editing-broken-migration/
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
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

Auto attendant not forwarding calls to Receiption

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