Cme Sip Configv3 Oct 2007

Cisco Unified Communications Manager
Express SIP Implementation Guide

Version 3.0
October 15, 2007
Revision History
Revision
1.0
2.0
Date
October 15, 2007
October 23, 2007
Author
Tony Huynh
Tony Huynh
3.0
October 30, 2007
Tony Huynh
Comments
Initial Draft
Added mwi-server
config
General formatting
Table of Contents
1.
Introduction...............................................................................................................................................3
1.1
Scope.................................................................................................................................................3
1.2
Audience ...........................................................................................................................................3
2. Overview ....................................................................................................................................................3
2.1
Hardware ...........................................................................................................................................3
2.2
IP Phone Firmware............................................................................................................................3
2.3
Network Topology, IP Addressing, and Extension Number Scheme ...............................................4
3. Configure System Configurations ...........................................................................................................5
3.1
Configure DHCP...............................................................................................................................5
3.2
Configure Fasthernet interface and switchports................................................................................6
3.3
Configure TFTP statements ..............................................................................................................8
4. Configure Basic Telephony Features ......................................................................................................8
4.1
Configure Voice Service Parameters ................................................................................................8
4.2
Configure Voice Register Global Parameters ...................................................................................8
4.3
Configure Connection to CUE ..........................................................................................................9
4.4
Configure Extension and Parameters ..............................................................................................10
4.5
Configure SIP Phone.......................................................................................................................11
4.6
Configure Advanced Parameters.....................................................................................................11
4.6.1
4.6.2
4.6.3
5.
4.7
Configure Parallel Hunt Group .......................................................................................................13
Troubleshooting ......................................................................................................................................14
5.1
SIP IP Phone does not get dial tone ................................................................................................14
5.1.1
5.2
5.2.1
5.2.2
5.3
5.3.1
5.4
5.4.1
6.
7.
Enable Presence for internal lines............................................................................................................ 11

Enable a Directory Number to be watched............................................................................................... 12
Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists......................................... 12
Steps to remedy......................................................................................................................................... 14
IP Phone cant upgrade to latest firmware ......................................................................................14

Steps to remedy......................................................................................................................................... 14
Debugs to collect ...................................................................................................................................... 14
Cant provision phone .....................................................................................................................14

Steps to remedy......................................................................................................................................... 14
SIP Phone gets DHCP but cant get communicate with CME........................................................15
Steps to remedy......................................................................................................................................... 15
Sample Config .........................................................................................................................................15

Reference .................................................................................................................................................22
7.1
Information on 3911 SIP IP Phones................................................................................................22
7.2
CME configuration guide................................................................................................................22
7.3
CME Documentation ......................................................................................................................22
1. Introduction
1.1 Scope
This document provides step-by-step instructions for setting up a standalone CME using SIP
phones. The document outlines a Cisco Unified Communications Manager Express (CME)
system with 4 SIP phones, with configurations for setting up the CME system and SIP phones.
Though the document will cover configuration steps to allow CME to interoperate with CUE,
CUE configuration is outside of the scope of this paper.
1.2 Audience
This document is provided for Cisco Systems Engineers (SEs) and resellers who wish to deploy
a CME system using SIP phones.
2. Overview
2.1 Hardware
The information in this document is based on the following hardware:
Cisco 2801 running CME 4.2 with IOS 12.4(11)XW2
9-port double-wide 10/100BASE-T Ethernet switch HWIC
Cisco 7970 SIP Phones
Cisco 3911 SIP Phone
2.2 IP Phone Firmware
Please refer to the following link to determine the appropriate SIP firmware to use for each
CME version.
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00805
acf50.html
Since we will be using CME 4.2, refer to the CME 4.2 Specifications link. In the following
deployment, the following phoneloads were used.
7970 IP Phone (SIP) SIP70.8-2-1S

3911 IP Phone (SIP) - SIP3951.8-0-2-9
You will need to have your CCO username/password in order to download SIP phoneloads.
The CME zip file does not have SIP phone loads, so you will have to procure the firmware at
the following site.
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-3900ser
For 7970 phone

Download SIP phone load cmterm-7970_7971-sip.8-2-1.zip file and unzip the file into your
TFTP folder.
For 3911 phone
Download the cmterm-3951-sip.8-0-2.zip file and unzip the file into your TFTP folder. NOTE:
3911 and 3951 phone use the same firmware, so dont be concerned that the filename reflects a
3951 phone.
After you have unzipped both the ZIP files in your TFTP folder, copy all the firmware files onto
the CME flash using your TFTP server. Make sure you copy all the following files onto flash.
SIP3951.8-0-2-9.loads
SIP3951.8-0-2-9.zz
DSP3951.0-0-0-1.zz
BOOT3951.0-0-0-9.zz
SIP70.8-2-1S.loads
term70.default.loads
term71.default.loads
apps70.8-0-2-55.sbn
cnu70.8-2-0-55.sbn
cvm70.sip.8-2-0-55.sbn
dsp70.8-2-0-55.sbn
jar70.sip.8-0-2-25.sbn
2.3 Network Topology, IP Addressing, and Extension Number Scheme

The following network topology will be used to deploy the CME system with SIP phones. The
table below outlines the network addressing scheme that will be used in this setup.
Purpose
Voice
Data
VLAN
192
100
Protocol
Phone Type
SIP
SIP
SIP
SIP
Voicemail Pilot
MWI On
Network
192.168.10.0/24
10.10.10.0/24
Interface
VLAN 192
VLAN 100
Extension
Number
7970
7970
3911
3911
100 AA Pilot
800 MWI Off
101
102
103
104
Interface Address
192.168.10.1/24
10.10.10.1/24
Phone Number
External Mask
4085251001
4085251002
4085251003
4085251004
110
801
Cisco Unified Communications Manager

Express SIP Network Topology
VOICE VLAN
DATA VLAN
PSTN
Communications
Manager Express
PILOT - 180
SIP Phones
7970
101
102
7970
3911
103
3911
104
192.168.10.0/24
10.10.10.0/24
3. Configure System Configurations

3.1 Configure DHCP
It will be necessary to configure two separate DHCP pools - IP Phones will use the Voice
DHCP pool and PCs will use the Data DHCP pool. IP Phones will need to use DHCP option
150 to provide the IP address of the TFTP Server.
If there are any devices in either pool with static IP addresses, make sure these addresses are
excluded from the DHCP pool to avoid addressing conflicts. You can use the show ip dhcp
binding command to verify which addresses the IP Phones and PCs are receiving from the
router.
Sample config:
ip dhcp excluded-address 10.10.10.1 10.10.10.10
!
ip dhcp pool data
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
!
ip dhcp pool voice
network 192.168.10.0 255.255.255.0
option 150 ip 192.168.10.1
3.2 Configure Fasthernet interface and switchports

In the following section, you will configure the VLAN interfaces for both the DATA and
VOICE VLAN and assign switchports into their respective VLANs.
NOTE: Prior to configuring VLANs, be sure to add the VLANs above to the VLAN database
with the following commands:
CME-SIP#vlan database
% Warning: It is recommended to configure VLAN from config mode,
as VLAN database mode is being deprecated. Please consult user
documentation for configuring VTP/VLAN in config mode.
CME-SIP(vlan)#vlan 100
VLAN 100 modified:
CME-SIP(vlan)#vlan 192
VLAN 192 modified:
CME-SIP(vlan)#exit
APPLY completed.
Exiting....
CME-SIP#
In the following section, configure the switchports to be connected to both the Voice and Data
VLANs. IP Phones will automatically be assigned into the Voice VLAN and PCs connected to
either the switchport directly or connected to the switchport on the IP Phone will be assigned to
the Data VLAN.
interface FastEthernet0/3/0
description 7970 Phone
switchport trunk native vlan 100
switchport mode trunk
switchport voice vlan 192
spanning-tree portfast
!
!
!
!!
interface Vlan100
description Data VLAN
ip address 10.10.10.1 255.255.255.0
!
interface Vlan192
description Voice VLAN
ip address 192.168.10.1 255.255.255.0
3.3 Configure TFTP statements

The following configuration allows CME to serve the IP Phones their firmware. [Required]
tftp-server flash:SIP3951.8-0-2-9.loads
tftp-server flash:SIP3951.8-0-2-9.zz
tftp-server flash:DSP3951.0-0-0-1.zz
tftp-server flash:BOOT3951.0-0-0-9.zz
tftp-server flash:SIP70.8-2-1S.loads
tftp-server flash:term70.default.loads
tftp-server flash:apps70.8-0-2-55.sbn
tftp-server flash:cnu70.8-2-0-55.sbn
tftp-server flash:cvm70.sip.8-2-0-55.sbn
tftp-server flash:dsp70.8-2-0-55.sbn
tftp-server flash:jar70.sip.8-0-2-25.sbn
4. Configure Basic Telephony Features

4.1 Configure Voice Service Parameters
Configure system to allow calls from SIP to SIP endpoints and enable SIP registrar. [Required]
voice service voip
allow-connections sip to sip Enable sip to sip calls
sip
registrar server expires max 1200 min 300 Enable IOS SIP registrar
4.2 Configure Voice Register Global Parameters

In the following section, you will be configuring voice register global parameters. Voice
Register Global configurations for SIP are similar to telephony-service configuration parameters
for SCCP phones. [Required]
Sample configuration:
voice register global
mode cme Set IOS SIP registrar to CME mode
source-address 192.168.10.1 port 5060 Set source address for phone registration
max-dn 20 Set max extensions
max-pool 10 Set max phones
load 7970 SIP7 SIP70.8-2-1S Specify phone loads for each phone type
load 3911 SIP3951.8-0-2-9 Specify phone loads for each phone type
authenticate register Set authentication for phone registration
authenticate realm cisco.com
tftp-path flash: Specify path for tftp files
create profile Create config files for all phones
dialplan-pattern 1 4085251... extension-length 3 Configure dial-plan pattern for system
4.3 Configure Connection to CUE

Configure a dial-peer and MWI server to interoperate with CUE. In order for CME to
interoperate with CUE, it is necessary to configure SIP CME as a back to back user agent
(B2BUA) - meaning all signaling and RTP stream goes through the CME. The following
configuration required is to enable connectivity to CUE.
dial-peer voice 2 voip
destination-pattern 1.0 Specify destination-pattern to reach CUE VM and AA
session target ipv4:10.1.10.1 Configure IP address to reach CUE
session protocol sipv2
dtmf-relay sip-notify Configure DTMF method to communicate with CUE
b2bua Enable B2BUA for CME for calls to CUE
codec g711ulaw
no vad
4.4 Configure MWI Server for SIP Phones

Configure MWI Server for SIP phones
sip-ua
mwi-server ipv4:10.1.10.1 expires 120 port 5060 transport udp

NOTE : 10.1.10.1 is the IP address of CUE
4.5 Configure Extension and Parameters

Configure voice register dn to create extension numbers for ephones. In the network topology
above, there are 4 extensions, which we will create now. [Required]
voice register dn 1
name Phone1 Set display name
label 4085251001 Set display label
number 101 Set extension number
mwi Enable MWI notification
call-forward b2bua noan 100 timeout 20 Configure call forward noan to voicemail pilot
call-forward b2bua busy 100 timeout 20 Configure call forward busy to voicemail pilot
allow watch Allow this number to be watched (presence)

!
voice register dn 2
name Phone2
label 4085251002
number 102
allow watch
!
voice register dn 3
name Phone3
label 4085251003
number 103
allow watch
!
voice register dn 4
name Phone4
label 4085251004
number 104
allow watch
4.6 Configure SIP Phone

In the following configuration, you will be configuring voice register pool parameters for each sip
phone. Voice register pool for SIP phones is identical to ephones for SCCP phones. [Required]
Sample configuration:
voice register pool 3
id mac 001A.A11B.500E Specify phone mac-address
type 3911 Specify phone type
number 1 dn 3 Assign button 1 dn tag 3
dtmf-relay sip-notify Configure dtmf-relay sip-notify to work with CUE
codec g711ulaw Specify codec
username user1 password cisco Configure username and password for SIP registrar
!
NOTE: Multiple methods for DTMF can be configured under voice register pool, but
for each SIP phone that has a voicemail box on CUE, configure dtmf-relay sipnotify.
4.7 Configure Advanced Parameters
In the following section, you will be configuring advanced parameters for SIP phones such as
presence with BLF status. Presence with BLF allows either a SCCP phone or SIP phone to
monitor the status of another SIP extension enabling presence information between phones.
[Optional]
The following phones support SIP presence service on CME.
Restrictions
BLF Call-List
Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
BLF Speed-Dial
Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
4.7.1 Enable Presence for internal lines
Perform the following steps to enable the router to accept incoming presence requests from
internal watchers and SIP trunks.
1.
2.
3.
4.
5.
6.
7.
8.
enable
configure terminal
sip-ua
presence enable
exit
presence
max-subscription number
presence call-list
9. end
Sample config
Presence Enable presence service
presence call-list Enable BLF monitoring of directory numbers
max-subscription 120 Configure max number watched sessions
!
sip-ua
presence enable Enable router to accept incoming presence request
4.7.2 Enable a Directory Number to be watched

To enable a line associated with a directory number to be monitored by a phone registered to a
Cisco Unified CME router, perform the following steps. The line is enabled as a presentity and
phones can subscribe to its line status through BLF call-list and BLG speed-dial features. There
is no restriction on the type of phone that can have its lines monitored; any line on any IP phone
or on an analog phone on supported voice gateways can be a presentity.
1.
2.
3.
4.
5.
6.
enable
configure terminal
voice register dn dn-tag
number number
allow watch
end
Sample Config
voice register dn 1
number 101
allow watch Allow this number to be watched
mwi
name Phone1
label 4085251001
- Repeat this configuration for each extension number that needs to be watched
NOTE: This step was already done when we first configured voice register dns.
4.7.3 Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory
numbers (presentities) through the BLF speed-dial and BLF call-list presence features. To
enable the BLF notification features on a SIP phone, perform the following steps.
1.
2.
3.
4.
enable
configure terminal
voice register pool pool-tag
number tag dn dn-tag
5. blf-speed-dial tag number label string

6. presence call-list
7. exit
8. voice register global
9. mode cme
10. create profile
11. restart
12. end
Sample Config
id mac 0016.47CD.9BD7
type 7970
number 1 dn 1
presence call-list Enable this phone to have presence call list
dtmf-relay sip-notify
username user1 password cisco
codec g711ulaw
blf-speed-dial 2 102 label "Phone2" Enable this line to monitor extension 1002
blf-speed-dial 3 103 label "3911-1" Enable this line to monitor extension 1003
blf-speed-dial 4 104 label "3911-2" Enable this line to monitor extension 1004
Note: Be sure to perform restart every time you change a SIP phone configuration.
Reference for configuring SIP Presence Service:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapt
er09186a008087793a.html#wp999282
4.8
Configure Parallel Hunt Group
In this section, will be assigning extensions 102, 103, and 104 into a parallel hunt group a parallel
hunt group is a hunt group that rings all members in the group simultaneously.
voice hunt-group 1
pilot 180 Configure Hunt group pilot number
list 102, 103, 104 Specify members in hunt-group
final 100 Specify final number as Voicemail Pilot
5. Troubleshooting
5.1
SIP IP Phone does not get dial tone
A common cause for SIP IP Phones not being able to get dial tone is that there is another phone
with the same extension as of CME 4.2, shared line is not supported on SIP Phones. Thus SIP
phones can not share the same extension among multiple phones. Additionally, make sure that the
SIP phone is provisioned with a proper extension.
5.1.1
5.2
Steps to remedy
Make sure that SIP phone has extension configured and extension shows on SIP
phone
Make sure that there is not another SIP or SCCP phone configured with the same
extension
IP Phone cant upgrade to latest firmware
The most likely causes for failure to be able to upgrade a phone is missing firmware files placed on
the CME flash or missing tftp-server commands.
5.2.1
Steps to remedy
First, check that the necessary firmware files are stored on the flash perform dir
flash to check flash for files
Check to see if you have updated the OS79XX.TXT file to reflect the correct
firmware the 79XX phones check this file to load the appropriate firmware and
change from SCCP to SIP
Check to make sure that the correct tftp-server statements are added for each
firmware file see section 3.3
Make sure the load command under voice register global is added for each type of
SIP phone section 4.2
5.2.2 Debugs to collect

In order to troubleshoot further, collect the following debugs in order to see if the phone is able
to get the appropriate phone loads from the CME flash.
5.3
Debug tftp events
Cant provision phone
The most like causes for not being able to provision is phone is that the phone doesnt have the
proper IP address with TFTP server option.
5.3.1
Steps to remedy
First, check to see that the phone receives an IP address and the proper TFTP server IP
address
Make sure that all proper voice register global commands are added see section 4.2.
5.4
Check that you are using the correct mac-address underneath each voice register pool
configuration
SIP Phone gets DHCP but cant get communicate with CME
The most likely cause for the SIP phone not being able to communicate with the CME after the
phone gets DHCP (with the proper TFTP server address) is because the username and password
credentials are not configured.
5.4.1
Steps to remedy
Check and make sure that username and password are defined under the voice
register pool configuration for each phone.
If the username and password are configured, but the command create profile is
not performed under voice register global, then the configuration file will not be
updated and the phone will not get this new configuration. Be sure to reset of the phone
after performing create profile.
6. Sample Config
CME-SIP#sh ver
Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.4(11)XW2, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2007 by Cisco Systems, Inc.
Compiled Mon 02-Jul-07 19:10 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T6, RELEASE SOFTWARE (fc1)
CME-SIP uptime is 18 hours, 55 minutes
System returned to ROM by reload at 17:01:34 UTC Wed Oct 3 2007
System image file is "flash:c2801-ipvoice-mz.124-11.XW2.bin"
Cisco 2801 (revision 4.1) with 235520K/26624K bytes of memory.
Processor board ID FHK084510HS
11 FastEthernet interfaces
1 terminal line
2 Voice FXO interfaces
3 DSPs, 48 Voice resources
1 cisco service engine(s)
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
CME-SIP# sh run
Building configuration...
Current configuration : 6227 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME-SIP
!
boot-start-marker
boot-end-marker
!
logging buffered 999999
no logging console
enable password cisco
!
no aaa new-model
ip cef
!
!
no ip dhcp use vrf connected
!
ip dhcp pool data
network 10.10.10.0 255.255.255.0
!
ip dhcp pool voice
network 192.168.10.0 255.255.255.0
option 150 ip 192.168.10.1
!
!
no ip domain lookup
multilink bundle-name authenticated
!
!
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 1200 min 300
!
!
!
!
!
voice register global
mode cme
source-address 192.168.10.1 port 5060
max-dn 20
max-pool 10
load 7970 SIP70.8-2-1S
load 3911 SIP3951.8-0-2-9
authenticate register
authenticate realm cisco.com
voicemail 100
tftp-path flash:
create profile sync 0000589556325309
!
voice register dn 1
number 101
mwi
call-forward b2bua noan 100 timeout 20
allow watch
name Phone1
label 4085251001
!
voice register dn 2
number 102
mwi
allow watch
name Phone2
label 4085251002
!
voice register dn 3
number 103
mwi
allow watch
name Phone3
label 4085251003
!
voice register dn 4
number 104
mwi
allow watch
name Phone4
label 4085251004
!
id mac 0016.47CD.9BD7
type 7970
number 1 dn 1
presence call-list
codec g711ulaw
blf-speed-dial 2 102 label "Phone2"
blf-speed-dial 3 103 label "3911-1"
blf-speed-dial 4 104 label "3911-2"
!
id mac 0014.6948.1D52
type 7970
number 1 dn 2
codec g711ulaw
!
id mac 001A.A11B.4FCE
type 3911
number 1 dn 3
codec g711ulaw
!
id mac 001A.A11B.500E
type 3911
number 1 dn 4
codec g711ulaw
!
voice hunt-group 1 parallel
final 100
list 102,103,104
pilot 180
!
!
!
!
voice-card 0
!
!
!
archive
log config
hidekeys
!
!
!
interface Loopback0
ip address 10.1.10.2 255.255.255.0
!
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface Service-Engine0/0
ip unnumbered Loopback0
service-module ip address 10.1.10.1 255.255.255.0
service-module ip default-gateway 10.1.10.2
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!

!
!
!
!
description Phone
!
description Phone
!
description Phone
switchport access vlan 192
!
description Phone
!

!
interface Vlan1
no ip address
!
interface Vlan100
ip address 10.10.10.1 255.255.255.0
!
interface Vlan192
ip address 192.168.10.1 255.255.255.0
!
ip route 10.1.10.1 255.255.255.255 Service-Engine0/0
!
!
ip http server
!
!
!
tftp-server flash:BOOT3951.0-0-0-9.zz
tftp-server flash:SIP3951.8-0-2-9.zz
tftp-server flash:DSP3951.0-0-0-1.zz
tftp-server flash:SIP3951.8-0-2-9.loads
tftp-server flash:SIP70.8-2-1S.loads
tftp-server flash:apps70.8-0-2-55.sbn
tftp-server flash:cnu70.8-2-0-55.sbn
tftp-server flash:cvm70.sip.8-2-0-55.sbn
tftp-server flash:dsp70.8-2-0-55.sbn
tftp-server flash:jar70.sip.8-0-2-25.sbn
!
control-plane
!
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
!
!
!
!
description ** cue voicemail pilot number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 100
b2bua
session target ipv4:10.1.10.1
codec g711ulaw
no vad
!
description ** cue auto attendant number **

translation-profile outgoing PSTN_CallForwarding
destination-pattern 110
b2bua
session target ipv4:10.1.10.1
codec g711ulaw
no vad
!
!
presence
presence call-list
max-subscription 120
!
sip-ua
mwi-server ipv4:10.1.10.1 expires 120 port 5060 transport udp
presence enable
!
!
!
!
!
line con 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password cisco
login
!
scheduler allocate 20000 1000
end
CME-SIP#
7. Reference
7.1
Information on 3911 SIP IP Phones
http://www.cisco.com/en/US/products/ps7193/products_qanda_item0900aecd8069bb1a.shtml
7.2
CME configuration guide
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_book091
86a00807c5776.html
7.3
CME Documentation
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186
a0080189132.html

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Cisco Unified Communications Manager

Express SIP Implementation Guide

October 30, 2007

Enable Presence for internal lines............................................................................................................ 11

IP Phone cant upgrade to latest firmware ......................................................................................14

Cant provision phone .....................................................................................................................14

Sample Config .........................................................................................................................................15

7970 IP Phone (SIP) SIP70.8-2-1S

For 7970 phone

2.3 Network Topology, IP Addressing, and Extension Number Scheme

Cisco Unified Communications Manager

3. Configure System Configurations

3.2 Configure Fasthernet interface and switchports

3.3 Configure TFTP statements

4. Configure Basic Telephony Features

4.2 Configure Voice Register Global Parameters

4.3 Configure Connection to CUE

4.4 Configure MWI Server for SIP Phones

mwi-server ipv4:10.1.10.1 expires 120 port 5060 transport udp

4.5 Configure Extension and Parameters

allow watch Allow this number to be watched (presence)

4.6 Configure SIP Phone

4.7.2 Enable a Directory Number to be watched

5. blf-speed-dial tag number label string

Configure Parallel Hunt Group

SIP IP Phone does not get dial tone

IP Phone cant upgrade to latest firmware

5.2.2 Debugs to collect

Debug tftp events

Cant provision phone

switchport voice vlan 192

switchport access vlan 192

description ** cue auto attendant number **

Information on 3911 SIP IP Phones

CME configuration guide

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description cue auto attendant number