R1.8B TSD
AT&T IP Flexible Reach
Cisco Unified Communications Manager Express
Configuration Guide
Issue 1.8
2/4/2008
TABLE OF CONTENTS
1 Introduction 3
2 Special Notes 3
3 Overview 4
4 Configuration Guide 6
4.1 Cisco Unified Communications Manager Express Configuration 7
4.1.1 Cisco Unified Communications Manager Express Version 7
4.1.2 Configuration for Cisco Unified Communications Manager Express to AT&T Calls 9
4.1.3 Configuration AT&T to Cisco Unified Communications Manager Express Calls 11
4.1.4 Fax Extensions 12
4.1.5 Interfaces 12
4.1.6 Connection to Unity Express 13
4.1.7 Calling Party Number Translation Configuration 15
4.1.8 Incoming Call Routing on Telephone Number 15
4.1.9 Codec 16
4.1.10 Cisco Unified Communications Manager Express Parameters (Conference, Transfer, etc) 16
4.1.11 Transcoder 17
4.1.12 Phone Configuration 17
4.1.13 VoIP Configuration 18
4.2 Cisco Unity Express Configuration 20
4.2.1 Unity Express Version 20
4.2.2 Unity Express Basic Parameters 20
4.2.3 Cisco Unity Express Application Configuration 22
4.2.4 Cisco Unity Express SIP and Other Parameters 23
4.2.5 Cisco Unity Express Phone Number to Application Assignment 24
4.2.6 Cisco Unity Express Mailbox Configuration 24
Introduction
This document provides a configuration guide to assist Cisco Unified Communications Manager Express administrators in connecting to AT&T IP Flexible Reach Services.
Special Notes
Emergency 911/E911 Services Limitations
While AT&T IP Flexible Reach services support E911/911 calling capabilities in certain circumstances, there are significant limitations on how these capabilities are delivered. Please review the AT&T IP Flexible Reach Service Guide in detail to understand these limitations and restrictions.
Calling Number Restricted / Privacy not supported
For calls from Cisco Unified Communications Manager Express to AT&T, the calling party number cannot be marked as restricted. This function is not supported.
Overview
The Cisco Unified CallManager Express environment is shown next.
[pic]
The Cisco Unified Communications Manager Express site consists of the following components.
• IP phones (customer managed) – These are Cisco Unified IP 7960G IP phones running the Cisco Skinny (SCCP) Protocol. Cisco SIP phones were not tested and are not supported.
• Cisco Unified Communications Manager Express (customer managed) – This is the Cisco Unified Communications Manager Express system with the Cisco Unity Express voice mail and auto attendant on an NM card. The key components in this configuration are shown next. FXS cards are used for fax. The DSP card is used for transcoding streams from the CUE (which uses G.711) and the WAN link to the AT&T network (which is configured for G.729B).
|Product Number |Product Description |
|CISCO2821-CCME/K9 |2821 Voice Bundle w/ PVDM2-32,FL-CCME-48,SP Serv,64F/256D |
|NM-CUE |Cisco Unity Express Network Module (includes SCUE-12-VM) |
|SCUE-LIC-12CME |Unity Express License 12 Voice Mailbox-Auto Attendant-CCME |
|VIC-4FXS/DID= |4 port FXS or DID VIC |
|PVDM2-64= |64-Channel Packet Voice/Fax DSP Module |
AT&T Managed Router (AT&T managed) – This is the router managed by AT&T. The router shall perform packet marking and QOS for voice. Cisco Unified Communications Manager Express uses a single IP address for signaling and a single IP address for media. The NAT router will support a single static NAT for the Cisco Unified Communications Manager Express signaling address and a single static NAT for the Cisco Unified Communications Manager Express media address.
Configuration Guide
This configuration guide specifies the Cisco Unified Communications Manager Express configuration that must be configured and updated to support the AT&T IP Flexible Reach.
This guide is not meant to be a stand-alone document, but as a complement to other Cisco-authored documentation. There are some excellent configuration examples and installation guides posted on Cisco’s Web Site:
1 Cisco Unified Communications Manager Express Configuration
1 Cisco Unified Communications Manager Express Version
For IP Flexible Reach, the Cisco Unified Communications Manager Express must be running one of the following IOS releases (i.e. 12.4(11)XJ or 12.4.15T1). You can run a “show version” command to check the version as shown below in the “system image file” lines.
CME-4_0_1-2821#sh ver
Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 12.4(11)XJ, REL
EASE SOFTWARE (fc1)
Synched to technology version 12.4(11)T
Technical Support:
Copyright (c) 1986-2006 by Cisco Systems, Inc.
Compiled Fri 22-Dec-06 01:06 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T7, RELEASE SOFTWARE (fc1)
CME-4_0_1-2821 uptime is 1 week, 5 days, 3 hours, 31 minutes
System returned to ROM by reload at 14:40:04 est Thu Jul 12 2007
System restarted at 14:41:47 est Thu Jul 12 2007
System image file is "flash:c2821nm-ipvoicek9-mz.124-11.XJ.bin"
CME41-CUE234-2821#sh vers
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 12.4(15)T
1, RELEASE SOFTWARE (fc2)
Technical Support:
Copyright (c) 1986-2007 by Cisco Systems, Inc.
Compiled Wed 18-Jul-07 06:21 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T7, RELEASE SOFTWARE (fc1)
CME41-CUE234-2821 uptime is 3 weeks, 4 days, 23 hours, 2 minutes
System returned to ROM by reload at 15:54:10 est Fri Nov 16 2007
System restarted at 15:56:50 est Fri Nov 16 2007
System image file is "flash:c2821nm-advipservicesk9-mz.124-15.T1.bin"
Please consult the following links to get additional information on CME and CUE.
Cisco Unified Communications Manager Express and SRST
Installing and Upgrading Cisco Unified CME Software
CUE - Software Download
Upgrading From the Previous Cisco Unity Express Release
Upgrading From an Earlier Cisco Unity Express Release
2 Configuration for Cisco Unified Communications Manager Express to AT&T Calls
This section describes the procedure for connecting to dual AT&T Border Elements. The actual AT&T border element IP addresses will be provided by AT&T Customer Care. The addresses, shown below, are examples.
There are 3 pairs of dial peers which perform the following functions.
• Dial peers 100 and 101 – These dial peers provide Cisco Unified Communications Manager Express to AT&T calling for 1+ 10 digit calls.
• Dial peers 200 and 201 – These dial peers provide Cisco Unified Communications Manager Express to AT&T calling for private dial calls and N11 calls.
• Dial peers 300 and 301 – These dial peers provide Cisco Unified Communications Manager Express to AT&T calling for 011+ international calls.
In each dial peer, the following parameters are configured.
• A translation profile (see later section) is referenced. For AT&T FlexReach Plan B (i.e. local option) calls, the calling number must be an AT&T provided DID. The translation profile maps the calling extension to the appropriate DID.
• One of the fax RTP payload types must be re-assigned so it does not conflict with the NTE (i.e. RFC4733/2833 telephone events) payload type typically used by AT&T.
• There is a reference to a voice class for the recommended codec order.
• DTMF relay is set to RTP-NTE (also known as RFC 4733/2833 telephone events).
• Fax rate is set to 14400 bps.
• The two IP addresses configured as session targets will be provided to the customer during test and turn-up of the AT&T IP Flex Reach service. The IP addresses used in the following configurations are used only as an example.
dial-peer voice 100 voip
description "SIP dialpeer to Production BE#1, 1 + 10"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern 1..........
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:135.25.29.135
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer voice 101 voip
description "SIP dialpeer to Production BE#2, 1 + 10"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern 1..........
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:12.120.205.133
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer voice 200 voip
description "SIP dialpeer to Production BE#1, Private dialplan"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern [2-9]T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:135.25.29.135
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer voice 201 voip
description "SIP dialpeer to Production BE#2, Private dialplan"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern [2-9]T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:12.120.205.133
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer voice 300 voip
description "SIP dialpeer to Production BE#1, International"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern 011T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:135.25.29.135
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer voice 301 voip
description "SIP dialpeer to Production BE#2, International"
translation-profile outgoing XlateCallingCalledNumber
preference 1
destination-pattern 011T
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type comfort-noise 13
voice-class codec 729
session protocol sipv2
session target ipv4:12.120.205.133
dtmf-relay rtp-nte
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
dial-peer hunt 1
Note 1: the “dial-peer hunt” command will cause the Cisco Unified Communications Manager Express to load-balance between the two Acme SBCs
3 Configuration AT&T to Cisco Unified Communications Manager Express Calls
The following dial peer is used for processing calls from the AT&T network to Cisco Unified Communications Manager Express. The key parameters in this dial peer are:
• One of the fax RTP payload types must be re-assigned so it does not conflict with the NTE (i.e. RFC4733/2833 telephone events) payload type typically used by AT&T.
• The incoming called-number command matches on the Cisco Unified Communications Manager Express extensions after the translation rule is applied (see later section). In this example, extensions start with the number “9”.
• DTMF relay is set to RTP-NTE (also known as RFC 4733/2833 telephone evenets).
• The codec must be hard coded to G.729B.
• Fax rate is set to 14400 bps.
dial-peer voice 400 voip
rtp payload-type cisco-codec-fax-ind 98
rtp payload-type nse 102
rtp payload-type nte 100
incoming called-number 9...
dtmf-relay rtp-nte
voice-class codec 729
fax rate 14400
ip qos dscp ef signaling
ip qos dscp ef media
!
4 Fax Extensions
The following dial peers are sample extensions used for fax.
dial-peer voice 500 pots
description "FXS Phone x9100"
destination-pattern 9100
port 0/1/0
!
dial-peer voice 501 pots
description "FXS Fax x9101"
destination-pattern 9101
port 0/1/1
!
dial-peer voice 502 pots
description "FXS Fax x9102"
destination-pattern 9101
port 0/1/2
!
dial-peer voice 503 pots
description "FXS Fax x9103"
destination-pattern 9101
port 0/1/3
!
5 Interfaces
This section provides examples of the IP interfaces required for the following VOIP functions. SIP signaling (i.e. control) and voice packets (i.e. media) are assigned to specific customer-provided IP addresses. These addresses must be provided to AT&T during provisioning as the customer signaling and media addresses. With this configuration, all media flows through the Cisco Unified Communications Manager Express. The IP address used in the following configuration is only as an example.
• Loopback interface for VOIP signaling (i.e. SIP signaling) - This is a customer-provided address that CallManager Express will use for SIP signalling to the AT&T IP Flexible Reach service. This must be a reachable address from the AT&T managed router on premises. The IP address used in the following configuration is only as an example.
• Loopback interface for VOIP media (i.e. voice packets) - This is a customer-provided address that CallManager Express will use for VOIP media to the AT&T IP Flexible Reach service. This must be a reachable address from the AT&T managed router on premises.
• Loopback interface for CME side of the CUE subnet.
• CUE interface (i.e. service-engine 1/0 interface)
Ensure that the command
ip route service-engine
is configured in order to establish a static route to the Cisco Unity Express module
interface Loopback1
description "VoIP Signaling"
ip address 172.16.5.248 255.255.255.255
!
interface Loopback2
description "VoIP Media"
ip address 172.16.5.249 255.255.255.255
!
interface Loopback3
description "Used for Service-Engine1/0"
ip address 10.1.10.2 255.255.255.0
interface Service-Engine1/0
description "Module for Cisco UnityExpress VoiceMail"
ip unnumbered Loopback3
service-module ip address 10.1.10.1 255.255.255.0
service-module ip default-gateway 10.1.10.2
no mop enabled
6 Connection to Unity Express
The following dial peers are used to match on the extensions used to connect to Unity Express.
• The first dial peer is used to connect to voice mail
• The second dial peer is used to connect to the auto attendant.
• The third dial peer is used to perform admin functions via telephone
Each dial peer has the following key parameters.
• The dialed extension for accessing the desired Unity Express application.
• The “b2bua” setting so that Cisco Unified Communications Manager Express acts as a back to back user agent on Unity Express calls.
• The IP address of the Unity Express.
• DTMF relay set to “SIP Notify”. Cisco Unified Communications Manager Express does the translation from RFC 4733/2833 to SIP Notify.
• Codec is set to G.711 ulaw. CME does the transcoding fro G.729 to G.711.
• No voice activity detection.
dial-peer voice 9999 voip
description "NM-CUE Unity Express VM"
destination-pattern 9999
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 9998 voip
description "NM-CUE Unity Express AA"
destination-pattern 9998
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 9997 voip
description "NM-CUE Unity Express Admin-Via-Telephone System"
destination-pattern 9997
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
7 Calling Party Number Translation Configuration
This section provides translation rule examples for mapping the calling extension to the AT&T provided DID number on calls from Cisco Unified Communications Manager Express to AT&T. This mapping is required to support FlexReach plan B which includes local services.
voice translation-rule 3
rule 1 /9000/ /7323682027/
rule 2 /9998/ /7323682021/
rule 3 /9999/ /7323680883/
rule 4 /9001/ /7322162708/
rule 5 /9002/ /7323680882/
rule 6 /9003/ /7323680883/
!
voice translation-profile XlateCallingCalledNumber
translate calling 3
Note 1: the customer will need to add a rule for each phone extension
8 Incoming Call Routing on Telephone Number
This section provides translation rule examples for mapping the called DID to the Cisco Unified Communications Manager Express extension on calls from AT&T to Cisco Unified Communications Manager Express.
Note that for virtual telephone numbers, AT&T will send 10 digits. For non virtual DIDs (as shown rule 5), AT&T will send 7 digits or less. The customer may request the exact number of digits that they want to receive.
voice translation-rule 2
rule 1 /7323682027/ /9000/
rule 2 /7323682021/ /9998/
rule 3 /7323680883/ /9998/
rule 4 /7322162708/ /9100/
rule 5 /0882/ /9002/
rule 6 /7323682020/ /9999/
!
voice translation-profile HoponXlations
translate called 2
!
voip-incoming translation-profile HoponXlations
Note 1: the customer will need to add a rule for each phone extension
9 Codec
The following voice class is used to out the codec preferences for calls from Cisco Unified Communications Manager Express to AT&T. The following list is recommended by AT&T.
voice class codec 729
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
10 Cisco Unified Communications Manager Express Parameters (Conference, Transfer, etc)
This section provides the key telephone service parameters (shown in bold) which are:
• DSP settings to point to DSP farm for transcoding, specify number of transcoder sessions. Note that MTP0013c45677d8” refers to the MAC address of the Cisco Unified Communications Manager Express inside interface.
• The load commands
• Maximum number 3 party conferences that are supported.
• Transfer parameters to support transfers to AT&T endpoints.
telephony-service
sdspfarm units 5
sdspfarm transcode sessions 16
sdspfarm tag 1 MTP0013c45677d8
load 7910 P00405000700
load 7960-7940 P0030702T023
load 7905 CP7905080001SCCP051117A
load 7912 CP7912080001SCCP051117A
max-ephones 25
max-dn 100
ip source-address 172.16.4.2 port 2000
auto assign 1 to 4
system message CME 4.1/CUE 2.3.3
voicemail 9999
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
web admin system name sysadmin secret 5 $1$Q2Br$zOuJHHoeW5cYpstYVXxGd0
web admin customer name custadmin secret 5 $1$I/ji$u8JEH1abgQDeoXmOyOz3E/
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 1..........
transfer-pattern .T
11 Transcoder
This section provides the commands required to configure the DSP farm for transcoding. Transcoding is used on 3 party conferences and calls to CUE. In both cases, G.729 is transcoded to G.711.
voice-card 0
dspfarm
dsp services dspfarm
sccp local GigabitEthernet0/0.2
sccp ccm 172.16.4.2 identifier 2821
sccp
!
sccp ccm group 1
associate ccm 2821 priority 1
associate profile 1 register MTP0013c45677d8
keepalive retries 5
!
dspfarm profile 1 transcode
codec g711ulaw
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
maximum sessions 4
associate application SCCP
Note 1: the command parameter “MTP0013c45677d8” refers to the MAC address of
The Cisco Unified Communications Manager Express inside interface:
CME-4_0_1-2821#sh int
GigabitEthernet0/0 is up, line protocol is up
Hardware is MV96340 Ethernet, address is 0013.c456.77d8 (bia 0013.c456.77d8)
12 Phone Configuration
This section provides a sample IP phone configuration. This example shows a phone with 3 lines. Key parameters for each line are:
• Rollover to voice mail on busy
• Rollover to voice mail after 10 seconds of no answer.
ephone-dn 4 dual-line
number 9003
label 9003-1
name CME x9003-1
call-forward busy 9999
call-forward noan 9999 timeout 10
ephone-dn 13
number 9003
label 9003-2
name CME x9003-2
call-forward busy 9999
call-forward noan 9999 timeout 10
ephone-dn 23
number 9003
label 9003-3
name CME x9003-3
call-forward busy 9999
call-forward noan 9999 timeout 10
ephone 4
username "jim"
mac-address 000F.9079.FD4C
type 7960
button 1:4 2:13 3:23
13 VoIP Configuration
The VOIP configuration provides commands for the following functions.
• The following SIP supplementary services are turned off: moved temporarily and refer.
• The IP fax protocol is set to G.711 ulaw. When fax is detected on a G.729 call, Cisco Unified Communications Manager Express supports the switch to G.711 for fax. This provides supports for G3 and SG3 fax machines.
• SIP signaling (i.e. control) and voice packets (i.e. media) is assigned to specific IP addresses. These addresses must be provided to AT&T during provisioning as the customer signaling and media addresses. With this configuration, all media flows through the Cisco Unified Communications Manager Express.
• The SIP user agent is set to send 2 invites and then fail over to an alternate dial peer.
voice service voip
media statistics
allow-connections h323 to sip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711ulaw
h323
sip
bind control source-interface Loopback1
bind media source-interface Loopback2
interface Loopback1
description "VoIP Signaling"
ip address 172.16.5.248 255.255.255.255
!
interface Loopback2
description "VoIP Media"
ip address 172.16.5.249 255.255.255.255
sip-ua
retry invite 2
retry options 0
2 Cisco Unity Express Configuration
This section provides some sample configurations for Cisco Unity Express. The Cisco Unity Express configuration for each customer will be different. Please go to to get the full Cisco Unity Express documentation.
1 Unity Express Version
For IP Flexible Reach, the Cisco Unity Express must be running one of the following releases (i.e. 2.3.3 or 2.3.4). You can run a “show version” command from the CUE prompt as shown below.
CME-4_0_1-2821# service-module service-engine 1/0 session
cue233-2821# sh software versions
Installed Packages:
Installer 2.2.1
Thirdparty 2.3.2
Bootloader (Primary) 2.1.2
Infrastructure 2.3.4
Global 2.3.3
GPL Infrastructure 2.3.0
Voice Mail 2.3.2
Bootloader (Secondary) 2.1.2
Core 2.3.4
Auto Attendant 2.3.0
Installed Languages:
English language pack 2.3.0
CME41-CUE234-2821#service-module service-engine 1/0 session
cue234-2821> sh software versions
Installed Packages:
- Installer 2.2.1
- Thirdparty 2.3.2
- Bootloader (Primary) 2.1.2
- Infrastructure 2.3.4
- Global 2.3.4
- GPL Infrastructure 2.3.0
- Voice Mail 2.3.2
- Bootloader (Secondary) 2.1.2
- Core 2.3.4
- Auto Attendant 2.3.0
Installed Languages:
- English language pack 2.3.0
2 Unity Express Basic Parameters
This section provides a brief description of some basic Cisco Unity Express parameters. Some of these basic parameters are:
• Configuration of group names, users and associated privileges
• Designation of opened and closed days as well as holidays.
cue233-2821# sh conf
clock timezone America/New_York
hostname cue233-2821
ip domain-name cust1.cme.
ip name-server 135.21.75.251 135.43.112.250
ntp server 135.25.29.14
software download server url "" credentials hidden "H/jo59iYw
jhJBrJR5fpHGwQMYy9J6PPbSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfG
WTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP"
log trace local enable
groupname Administrators create
groupname Broadcasters create
username cueadmin create
username jim create
username mgarrish create
username lthomas create
username rutano create
username jim phonenumber "9003"
username mgarrish phonenumber "9000"
username lthomas phonenumber "9001"
username rutano phonenumber "9002"
groupname Administrators member cueadmin
groupname Administrators member jim
groupname Administrators member mgarrish
groupname Administrators privilege superuser
groupname Administrators privilege ManagePrompts
groupname Administrators privilege broadcast
groupname Administrators privilege local-broadcast
groupname Administrators privilege ManagePublicList
groupname Administrators privilege ViewPrivateList
groupname Administrators privilege vm-imap
groupname Broadcasters privilege broadcast
restriction msg-notification min-digits 1
restriction msg-notification max-digits 30
restriction msg-notification dial-string preference 1 pattern * allowed
backup server url "" credentials hidden "GixGRq8cUmFqr
OHVxftjAknfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYH
fmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP"
calendar biz-schedule systemschedule
closed day 1 from 00:00 to 24:00
open day 2 from 08:00 to 18:00
open day 3 from 08:00 to 18:00
open day 4 from 08:00 to 18:00
open day 5 from 08:00 to 18:00
open day 6 from 08:00 to 18:00
closed day 7 from 00:00 to 24:00
end schedule
calendar holiday date 2006 12 25 description "Christmas"
calendar holiday date 2006 01 01 description "New Year's Day"
calendar holiday date 2007 01 01 description "New Year's Day"
calendar holiday date 2007 12 25 description "Christmas"
calendar holiday date 2007 02 04 description "Super Bowl XLI"
3 Cisco Unity Express Application Configuration
This section provides some example configurations for some basic applications. Each of these applications contain an associated script. Please consult the CUE documentation at for script creation instructions. These applications are:
• Auto attendant
• Message Waiting Indication
• Message Notification
• Prompt management (ability to manage the Cisco Unity Express via a phone interface).
• Voice mail
ccn application autoattendant
description "autoattendant"
enabled
maxsessions 4
script "aa.aef"
parameter "busOpenPrompt" "AABusinessOpen.wav"
parameter "operExtn" "9000"
parameter "welcomePrompt" "AAWelcome.wav"
parameter "disconnectAfterMenu" "false"
parameter "busClosedPrompt" "AABusinessClosed.wav"
parameter "allowExternalTransfers" "false"
parameter "holidayPrompt" "AAHolidayPrompt.wav"
parameter "businessSchedule" "systemschedule"
parameter "MaxRetry" "3"
end application
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 4
script "setmwi.aef"
parameter "CallControlGroupID" "0"
parameter "strMWI_OFF_DN" "8001"
parameter "strMWI_ON_DN" "8000"
end application
ccn application msgnotification
description "msgnotification"
enabled
maxsessions 4
script "msgnotify.aef"
parameter "logoutUri" ""
parameter "DelayBeforeSendDTMF" "1"
end application
ccn application promptmgmt
description "promptmgmt"
enabled
maxsessions 1
script "promptmgmt.aef"
end application
ccn application voicemail
description "voicemail"
enabled
maxsessions 4
script "voicebrowser.aef"
parameter "uri" ""
parameter "logoutUri" ""
end application
4 Cisco Unity Express SIP and Other Parameters
This section provides a brief description of some SIP and other Cisco Unity Express parameters. Some of these basic parameters are:
• IP address of the Cisco Unified Communications Manager Express(i.e. gateway IP address).
• Setting of the dtmf-relay to sip-notify.
ccn engine
end engine
ccn subsystem jtapi
ccm-manager address 0.0.0.0
end subsystem
ccn subsystem sip
gateway address "172.16.5.248"
dtmf-relay sip-notify
end subsystem
5 Cisco Unity Express Phone Number to Application Assignment
This section shows the assignment of phone extensions to Cisco Unity Express applications:
ccn trigger sip phonenumber 9997
application "promptmgmt"
enabled
locale "en_US"
maxsessions 1
end trigger
ccn trigger sip phonenumber 9998
application "autoattendant"
enabled
maxsessions 4
end trigger
ccn trigger sip phonenumber 9999
application "voicemail"
enabled
maxsessions 4
end trigger
service phone-authentication
end phone-authentication
service voiceview
enable
end voiceview
voicemail default language en_US
voicemail default mailboxsize 6000
voicemail broadcast recording time 300
voicemail mailbox owner "jim" size 6000
description "jamster mailbox"
end mailbox
6 Cisco Unity Express Mailbox Configuration
This section shows some sample phone mailbox configurations.
voicemail mailbox owner "lthomas" size 6000
description "lthomas mailbox"
end mailbox
voicemail mailbox owner "mgarrish" size 6000
description "mgarrish mailbox"
end mailbox
voicemail mailbox owner "rutano" size 6000
description "rutano mailbox"
end mailbox
end
cue233-2821#
This Customer Configuration Guide ("CCG") is offered as a convenience to AT&T's customers. The specifications and information regarding the product in this CCG are subject to change without notice. All statements, information, and recommendations in this CCG are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this CCG.
In no event shall AT&T or its suppliers be liable for any indirect, special, consequential, or incidental damages, including, without limitation, lost profits or loss or damage arising out of the use or inability to use this CCG, even if AT&T or its suppliers have been advised of the possibility of such damage.
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PSTN
Network
Gateway
Border
Element
IP Border
Element
AT&T
Managed
Router
With NAT
Switch
Public Side
Call Control
Element
AT&T Managed
With
Voice GW
Legacy Circuit
PBX
Private Side
Call Manager Express
Application
Server
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