Vuc SMB CP Cme

RouteHub Group, LLC Page 1 www.routehub.
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Deploying Cisco UC
Manager Express and
Unity Express
Voice & Unified Communications: Small Business

Practical Cisco Training for Network Engineers & Consultants!

RouteHub Group, LLC
www.RouteHub.net

June 30, 2010

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Table of Contents
1 Introduction 8
2 Concepts 9
3 Design 11
3.1 Our Design Small Business Voice Design 11
3.2 Requirements 13
3.3 Solutions and Topology 13
3.4 Topology Services and Sub-Services 14
3.5 Hardware & Software 15
3.6 Network Diagram 16
4 Configuration for CME 17
4.1 Initial Configuration 17
4.2 CallManager Express 18
4.2.1 Telephony Service (telephony-service) 18
4.2.2 Directory Number (DN) Configuration (ephone-dn) 23
4.2.3 IP Phone Configuration (ephone) 24
4.3 Voice and Data VLAN Configuration 25
4.4 Configuring DHCP on Cisco IOS 27
4.5 Mapping an analog line (DID) to an IP phone 27
4.6 Configuring FXS port as a SCCP port 30
4.7 CME as SIP Server for SIP Clients 32
4.8 Blocking incoming calls from PSTN 33
4.9 Setting up an Authenticated SIP Trunk to SIP Provider 35
4.10 Phone Directory 38
4.11 Single Number Reach (SNR) 38
4.12 Sending Calls to Voicemail (CUE) 39
4.13 Conferencing 41
4.13.1 MeetMe Conferencing 41
4.13.2 Ad-Hoc Conferencing 44
4.14 Paging 45
4.15 Personal Speed Dial 46
4.16 Upgrading CallManager Express 46
4.17 Intercom 47
4.18 Hunt Group 48
4.19 Call Park 48
4.20 How to setup Phone Softkey templates 49
4.21 Extension Mobility 50

4.22 How to setup a custom ring tone? 52
4.23 Call Center 55
4.24 Fax to Email using T.37 58
4.25 Phone Services 61
4.26 Cisco CME using Exchange 2007 UM 62
4.27 Using a XML Menu File for Phone Services 64
4.28 Installing SIP Firmware on Cisco 7940/7960 66
4.29 VoiceView 71
4.30 Installing SIP Firmware on Cisco 7941/7961/7962+ 80
4.31 Cisco Unified CallConnector 84
4.31.1 Server Installation 85
4.31.2 Components 109
4.31.3 Adding a new user 111
4.31.4 Client Installation 117
4.31.5 Using Cisco Unified CallConnector 124
5 Configuration for CUE 134
5.1 Access to CUE 134
5.1.1 CME Configuration 134
5.1.2 Console into the CUE Service Engine. 136
5.2 Unity Express 136
5.2.1 CUE Global Configuration 137
5.2.2 Enable Voicemail Services 138
5.2.3 Sending Calls to Voicemail on CME 139
5.2.4 Create User Voice Mailboxes 140
5.2.5 Enable other CUE services (like Auto Attendant) 141
5.3 Upgrade CUE to Version 7.x 142
5.4 Coping Files to CUE via CLI 145
5.5 Auto Attendant 146
5.6 Voicemail Email Notifications 149
5.7 CUE and CME on separate routers 151
5.7.1 CME Router 151
5.7.2 CUE Router (Cisco CUE Router Configuration) 154
5.7.3 CUE Router (CUE Configuration) 155
5.8 Live Record 158
5.9 Downgrade CUE software 160
5.10 Basic CUE Start-Up Wizard 167
6 Monitor 170
6.1 Operations 170
6.1.1 IP Phones 170
6.1.2 Conferencing and DSP resources 172
6.1.3 Dial Plan and Cisco CallManager Express 177

6.1.4 SIP 182
6.1.5 External Calling summary 191
6.1.6 Email Notification and Voice Messaging (CUE) 192
6.2 Troubleshooting 194
6.2.1 Root Causes 194
6.2.2 Initial questions to ask 194
6.2.3 Typical fixes 195
7 Sample Full Configuration 196
7.1 CME and CUE on UC520 196
7.1.1 CME 7.1 on UC520 196
7.1.2 CUE 7.0.1 on UC520 216

1 Introduction
Many site focused on providing training towards certifications or exams. These are important
for career development and we have CCIE, CCNP, and CCNA certifications. So we know
that they are very valuable to your network engineering career, however, they do not teach
practical network training relevant for network engineers and consultants in the real world.

This is what our training format is based upon providing practical solutions and technologies
that are deployed in real working environment. Our training workbooks provide the four
major components:

Concepts
Design
Configuration
Monitor

Learn the concepts that matter in terms of the components and protocols involved for a
technology's operation.

Learn how to design a network solution with practical steps, considerations, and tools for
your company or clients.

Learn how to configure a network with best practices and get operational step-by-step. We
also include full working configuration files for our workbooks.

Learn how to monitor, troubleshooting, and confirm the operational state of your configured
network.

All four are important for network engineers and consultants to know how to manage a
network in real time.

2 Concepts
A Cisco IP Telephony solution really has two categories when discussing which option is best
for a customer; Enterprise and Small Business. Enterprise Cisco IP Telephony would involve
the Cisco Communications Manager (previously known as Cisco CallManager) and Cisco
Unity for Unified Messaging. Small Business Cisco IP Telephony would involve CallManager
Express and Cisco Unity Express, taking full advantage of their existing Cisco router for other
network services (e.g. Internet connectivity, Firewall, and Remote Access capabilities such
as SSL VPN and Client-based IPSec VPN access).

Why would a Small Business use a Cisco solution? Well, many business owners are
cautious to place all their eggs in one basket in general, but companies know that Cisco is a
solid company with a strong networking focus to their businesses model. Now, many
products that Cisco has offered to their customers were really designed for medium to large
business due to capabilities with performance, security, and reliability aspects that are
important to a business. Well because of those capabilities a high price tag is associated
with it. Then again, you get what you pay for. I have been deploying Cisco networks for
more than 12 years and with Cisco hardware deployment (design and implemented correctly
of course), it just works! I rarely touch or revisit a Cisco product implementation for continued
support or troubleshooting. Hence, why many companies tend to choose Cisco for this
reason among other critical requirements needed in a solution.

Cisco's Enterprise IPT solution is tailored for environments, which require 2500 to 10,000 IP
endpoints. This would involve many Communication Manager Servers deployed in a cluster
setup, which is very clean and provides ease of administration. Tracing call activity with SDL
files for example can be a little tricky at times. For small business, going with the
Communication Manager product is an overkill, not needed, and very expensive. Small
Business tends to turn to NEC, for example, for a Small Business IP Telephony solution.

Another alternative with the Cisco Small Business product line is CallManager Express and
Cisco Unity Express. They provide scaled down capabilities of the Cisco Communication's
Manager product, but offer its full capabilities for call routing and voicemail for customers in a
single solution, not turnkey style!

Basically, as a consultant, the Cisco router must be setup and configured for CallManager
and Unity initially. Administrating IP Phones, extensions, etc can be done via the web portal
provided for the IP Telephony Express Suite (CallManager and Unity Express). I have done
deployments where we have a Cisco 2800 series router with IOS 12.4 running with CME
(CallManager Express) capabilities and a AIM-CUE (providing Cisco Unity Express) installed
plus a Wireless 802.11g WIC card to provide Wireless capabilities. With 12.4 we can setup
WebVPN or SSL VPN for users. We can setup DMVPN for hub-and-spoke VPN capabilities
to different sites and not compromising security. We can enable a stateful firewall on the
router that would connect to a DSL or Cable Modem device for Internet Access. To even
providing DHCP services for local LANs to Quality of Service (QoS) to preserve voice quality.
This is one design option for deploying Cisco CallManager Express and Unity Express to a
Small Business with those added capabilities. It becomes more cost efficient and it's robust.
So, that is the design (one way at least) for how you can use this solution.

However two issues arise, 1) what about Small and SMB sizes, and 2) Turn-key solution!
For organizations between 250 to 500 users, CallManager and Unity Express is not any
option any more. Therefore, a Cisco Communication Manager is required, hence high cost
and endless features going unused. Cisco in the past year started developing products and

solutions tailored for Small and SMB size customers. There is now a Cisco Communication
Manager appliance designed for Small Businesses in a single turnkey solution providing call
manager, voicemail, and other great functionality. This part of Cisco Communication
Manager 6 product line is the "Business Edition" and is designed to support up to 500 users
and endpoints. Another SMB turnkey solution is the Unified Communications 500 Series.
Remember what we discussed in terms of using a Cisco 2800 series for example and
integrating CME, CUE (via NM-CUE or AIM-CUE), Wireless, etc.

Well, I think Cisco has been reading all of our minds. The Unified Communications 500
device provides all of these options in one single turnkey solution. The UC500 can be
configured via CLI or through a new GUI application for easy administration. The choice is
up to you as a consultant or engineer. I have deployed numerous UC520 products to many
SOHO and Small size businesses providing a robust voice solution. Plus it's a lot quieter
than having a 2800 or higher running in your facility.

So, when it comes to designing an IPT solution choosing the right solution is based on the
size of the environment, growth considerations, and functionality required by the customer.
SOHO to Small networks would normally get CallManager Express, Unity Express, or the
Unified Communication 500 device. Small to SMB networks would be border line with
CallManager Express & Unity Express, but Cisco Communications Manager 6 Business
Edition would be a better fit especially when potential growth comes into play. And for
Medium to Large Enterprise, the full blown version of Cisco Communication Manager
deployed in clusters would be recommended.
In terms of the Cisco Communication Manager family series, there is CallManager 4,
Communication Manager 5 (previously called CallManager 5), and now Communication
Manager 6. Communication Manager 5 and 6 are pretty much the same except for a few
enhancements provided in the 6 release such as integrated Unified Presence within the
product than separating it out plus providing great Mobile capabilities allowing your cell
phone and office phone to ring at the same time.

3 Design
3.1 Our Design Small Business Voice Design

Our network design will include Voice and Unified Communication user services. Below are
the details on how our voice network is designed.

Requirements
Our network consist of 4 users with a potential growth up to 8 users (a Small Business) each
with an IP Phone, but our configuration will only show a handful of our phones plugged in.
Our voice network will use a single analog line from their local PSTN provider with a
dedicated phone number (aka DID number). The small office will require use of all basic
voice features such as conferencing, redial, speaker phone, etc.

Voicemail is also needed for our users and ideal if there voicemail messages can be sent via
email. Other unique features with the voice solution would also be great, but not required.
They only have a single location and will require some form of remote access to access the
office's resources remotely.

Some of the requirements and expectations include:
Voice system with voicemail application user services for the office
Remote access to resources

Solutions and Topology
Based on our requirements, our applicable solutions for our environment are the following:

LAN; required because each site has a LAN network where all servers, desktops,
and IP phones would connect into for access to other user services.

Our LAN topology, our hierarchical design, would be a single tier giving us a LAN Collapsed
Core model since our office consists of less than 24 devices.

Our general design with our solution will consist of a single LAN subnet using the following
schema: 10.67.78.Y /24 configured on our LAN Collapsed Core; where "Y" is designated for
the node.

Bandwidth services within our LAN will be FastEthernet.

Topology Services
Within our LAN topology we will utilize the following network services applicable to our
environment and requirements:

Required Services
Routing & Switching: Static routing; routing is required and since we do not have
multiple sites or routing devices we would use default route with the ISP.
Security (VPN): SSL VPN; provides remote access services for users to access the
office resources remotely using HTTPS/SSL VPN.

As for our user services, we will implement Voice and Unified Communications on our LAN
network with a call processing and voicemail solution to support up to 8 users. Using voice
requires our LAN for additional network services such as a Multicast, Quality of Service
(QoS), VLANs, and 802.1q Trunking.

Hardware
We need hardware to support our topology services, user services, bandwidth services, and
requirements in our design that includes the following:

Voice: Call Processing & Voicemail support up to 8 users with basic voice features
Multicast
Quality of Service (QoS)
Static Routing
SSL VPN
VLANs
802.1q Trunking
FXO port for the analog line

In our network we can consolidate our LAN Collapsed Core and our voice user services
together for simplicity.

The hardware chosen for our design will consist of the following:
Hardware: Cisco UC520 (license for 8 users) integrated with Voicemail service
engine
o Cisco UC Manager Express (using OS 7.x)
o Cisco Unity Express (using OS 7.x)
Software: IOS 12.4 Advanced IP Services Feature set
Cisco IP Communicator
Cisco IP 7970 phone
Third-Party SIP phone

Our Cisco UC520 appliance allows us to send voicemail messages via email to access with
other voice features to include some of the following:
Extension Mobility
Single Number Reach (SNR)
Live Record
Paging, Intercom
SIP Services
and more!

3.2 Requirements

First, we need to determine all the business and technical requirements. Understand what is
needed, the expectations involved, budgetary considerations, network services, security
regulations, and more much outlined by the company or business

We would gather details for building our design based on the following:
Requirements and Expectations
Traffic
Budgetary Considerations
Existing Components and Services
Technical Objectives

The technical objectives are what define best practices and recommendations in a network
design. These are often challenges that many networks face early or further down the road
with a network. When there are issues its usually due to one of the objectives that were no
met or considered during the design phase.

Below are the technical objectives our design should consider, include, and bring up with the
requirements gathering:
Performance
Reliability
Scalability
Security
Flexibility
Network Management

3.3 Solutions and Topology

Once the requirements and objectives have been gathered, that info will help with the design
process of our solutions and topology.

At a high level the solutions is the network that deals with a specific function or task based on
the requirements gathered. Many network solutions listed here do require the existing of
other solutions to work. The one network solution that is required for all solutions is the LAN
solution which is essentially the network backbone that connects all the other solutions
together.

Below are the solutions we can choose from.
Local Area Network (LAN), Wireless
Wide Area Network (WAN), Metropolitan Area Network (MAN)
Internet Edge
Data Center

Once the solutions have been determined it is time to build our topology. The topology is
basically the framework in our design that doesnt contain any technologies, services,
protocols, or hardware devices by name yet. We are essentially just building a street with
nothing on it.

There are many ways to build a design and usually common topologies and case studies are
often used.

These topologies really include tier levels in the design. One way to explain is with a LAN
topology which is often discussed in many networking textbooks. A best practice and
recommended LAN would consist of a LAN Core, LAN Distribution, and LAN Access. This is
a tier level model consisting of 3 tier levels, each with a certain ideal purpose.

A LAN Access provides direct access to nodes like computers, printers, IP Phones, access
points, etc. LAN Distribution deals with aggregating the traffic from the Access layer
including other roles with routing, switching, and security policies. And the LAN Core is seen
at the backbone where the LAN Distribution connects into providing high-speed switching
and forwarding. This three tier model accommodates much of the technical objectives
especially with scalability and reliability among others. But a 3-tier model is often seen with
larger networks.

Some solutions typically can have 1 or 2 tiers in most designs. Again 3 tier designs are often
seen with large size networks or very large networks. But some of the tier levels can be
consolidated where needed and the hardware that you choose that can also change the tier
level in the design. For example, an Internet Edge solution typically consists of 3 tiers (the
Edge Router, the Edge Switch, and the Perimeter Firewall). Well nowadays the edge switch
has been eliminated being integrated with the Edge Router leaving us with a 2 tier model,
which is the most common, however, the firewall services can also be integrated with our
Edge router that provide stateful firewall inspection with capabilities such as rACL (Reflexive
ACL) or CBAC. Thus, our Internet Edge device can be a 1 tier model.

2 tier models are very common for small and medium sized networks.

3.4 Topology Services and Sub-Services

Once the topology has been determined (or narrowed down), the next thing to determine is
the topology services that will overlay on-top of our topology.

This can include the following services:
Routing & Switching
Security & VPN
Tunneling
Voice & Unified Communications
Wireless
Other Technologies (like QoS and HSRP)

Topology sub-services deals with the extended features within the services within the
network design.

For example, one of our topology services could be Routing using OSPF. Well OSPF has
many design considerations and best practices that can include configuring route
summarization within a LAN Distribution to send summary routes up to a LAN Core. A

common best practice discussed with OSPF including Stub routing within the LAN Access
network among other sub-services.

For example, MPLS, which is a topology service, these are sub-services that can be
deployed with MPLS.
General
Route Reflectors
VRF Selection
Traffic Engineering (TE)
Extranet
MPLS over GRE, MPLS over DMVPN
QoS service to MPLS VPN
IPv6
Internet Access service
Multicast service to MPLS VPN

3.5 Hardware & Software

Determine the best hardware and software solutions for each component in the design to
accommodate the following points:
Requirements
Topology Service and Sub-Services
Business Size considerations

The hardware device can be any vendor besides Cisco. Make sure the hardware chosen
supports the requirements and services in our design including considerations for the
business size of the network and the technical objectives.

3.6 Network Diagram

Below is the network diagram showing our completed design with voice user services.

4 Configuration for CME

4.1 Initial Configuration

The first we need to do is console or connect into each device on our network based on the
information presented in the network diagram.

Second, complete all basic configurations for all devices based on the following:
Configure all interfaces based on the network diagram in terms of IP addressing and the
subnet mask.

Next enable all interfaces by issuing a no shutdown

Once that has been completed we need to check on two things.

First confirm that all interfaces are up and running. This command will show all interfaces
and there status in a basic or brief view. Confirm that all interfaces once configured shows
an UP UP status.
show ip interface brief

And second, confirm basic network connectivity by pinging the directed connected IP address
of the other router. Do this for each device.

4.2 CallManager Express

Cisco CallManager Express or Cisco Unified Communication Manager (UCM) Express is a
call processing solution aimed for Small/SMB businesses. They provide many voice features
and applications such as Call Center. Call Processing is the central component in a VoIP
network infrastructure where everything connects such as IP Phones, Voice Gateways, and
other external voice applications like voicemail.

Before any of the features below can be configured we need to enable Cisco CallManager
Express (CME) or Cisco Unified Communication Manager (UCM) Express on a Cisco router
supported for CME.

1. We will need to configure our CME server, which is done under telephony-service.
2. Next we need to configure our Directory numbers (DN), which is a unique extension
or number used by users with IP phones. This is done under what is called ephone-
dn.
3. And last we associate the configured DN to a physical IP Phone on the network.
This is done under what is called ephone.

4.2.1 Telephony Service (telephony-service)

To enable a basic configuration for CME is actually very simple. All the extra features added
is what makes the configuration look very long.

STEP 1: LAN INTERFACE AND IP CONFIGURATION SUMMARY
Our CME configuration will be on a Cisco UC520 appliance supporting up to 8 users. Our
LAN interface on the UC520 where all of our IP phones and systems are connected to is
configured as followed:

vlan 10
name ROUTEHUB-VLAN

interface FastEthernet0/1/1
description IP Phone Port
switchport access vlan 10

interface Vlan10
no ip address
no ip redirects
no ip unreachables
no ip proxy-arp
bridge-group 10
bridge-group 10 spanning-disabled

interface BVI10
ip address 10.67.78.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp


STEP 2: ENABLE TELEPHONY SERVICE
We will enable CME on our router using the IP address from our LAN and specify the SCCP
port number of 2000. We need to enable telephony-service first before enabling all other
commands within this section.

telephony-service
ip source-address 10.67.78.1 port 2000

STEP 3: TIMEOUT FOR INTER-DIGITS
When calls are placed from any IP phone registered with CME it will take 5 seconds for CME
to setup the call. This is important to configure since the default timeout value is long and is
a common compliant among users who are placing calls and it takes a long time for the call
to get connected.

timeouts interdigit 5

STEP 4: BANNER ON IP PHONE
We can configure a short banner that would be displayed on all IP phones near the softkeys.
In our configuration our banner would read RouteHub UC520.

system message RouteHub UC520

STEP 5: AUTO-REGISTRATION
We will use auto-registration on CME where any new phone plugged into the network will
automatically get a temporary DN from a list of DN configured on CME. In our configuration
our auto-registration will be the DN from profile 19.

auto assign 19 to 19

STEP 6: VIDEO SUPPORT
If video related services with Cisco VTAdvantage are used it can be enabled globally for IP
phones that support video capabilities with Cisco VTAdvantage like the Cisco 7970 and 7960
phone series.

video

STEP 7: TIMEZONE
Next we will specify the timezone that CME will refer and use for the time for all IP phones
connected to CME.

In our configuration we will choose 5 which is for PST.

time-zone 5

Below are the numbers for other time-zone numbers we can choose from:


1 Dateline Standard Time -720
2 Samoa Standard Time -660
3 Hawaiian Standard Time -600
4 Alaskan Standard/Daylight Time -540
5 Pacific Standard/Daylight Time -480
6 Mountain Standard/Daylight Time -420
7 US Mountain Standard Time -420
8 Central Standard/Daylight Time -360
9 Mexico Standard/Daylight Time -360
10 Canada Central Standard Time -360
11 SA Pacific Standard Time -300
12 Eastern Standard/Daylight Time -300
13 US Eastern Standard Time -300
14 Atlantic Standard/Daylight Time -240
15 SA Western Standard Time -240
16 Newfoundland Standard/Daylight Time -210
17 E. South America Standard/Daylight Time -180
18 SA Eastern Standard Time -180
19 Mid-Atlantic Standard/Daylight Time -120
20 Azores Standard/Daylight Time -60
21 GMT Standard/Daylight Time +0
22 Greenwich Standard Time +0
23 W. Europe Standard/Daylight Time +60
24 GTB Standard/Daylight Time +60
25 Egypt Standard/Daylight Time +60
26 E. Europe Standard/Daylight Time +60
27 Romance Standard/Daylight Time +120
28 Central Europe Standard/Daylight Time +120
29 South Africa Standard Time +120
30 Jerusalem Standard/Daylight Time +120
31 Saudi Arabia Standard Time +180
32 Russian Standard/Daylight Time +180
33 Iran Standard/Daylight Time +210
34 Caucasus Standard/Daylight Time +240
35 Arabian Standard Time +240
36 Afghanistan Standard Time +270
37 West Asia Standard Time +300
38 Ekaterinburg Standard Time +300
39 India Standard Time +330
40 Central Asia Standard Time +360
41 SE Asia Standard Time +420
42 China Standard/Daylight Time +480
43 Taipei Standard Time +480
44 Tokyo Standard Time +540
45 Cen. Australia Standard/Daylight Time +570
46 AUS Central Standard Time +570
47 E. Australia Standard Time +600
48 AUS Eastern Standard/Daylight Time +600
49 West Pacific Standard Time +600
50 Tasmania Standard/Daylight Time +600
51 Central Pacific Standard Time +660
52 Fiji Standard Time +720
53 New Zealand Standard/Daylight Time +720


STEP 8: VOICEMAIL
Next we will specify what the voicemail main number will be. This basically configures a
speed dial for users who want to check their voicemail they can simply press the voicemail
button on their phone.

voicemail 6000

STEP 9: WEB ADMIN ACCOUNT
We can also configure CME, our DNs, and phones directly from a GUI interface which is
enabled once CME is configured. But, we need to configure a username and password to
access the GUI page. We can configure our web admin account by doing the following:

web admin system name admin secret cisco123

To access the GUI page we would simply go to a web browser use the IP address of the
CME server followed by ccme.html (for example):

http://10.67.78.1/ccme.html

That will prompt for a username and password where we would input admin / cisco123.

Once we are logged in successfully we should see the following page where we can
configure our Phones, Extensions, and System Parameters listed under Configure among
other configuration.

STEP 10: MUSIC ON-HOLD (MOH)
MOH is also enabled by default and the following file is used for MOH, which is not required
to be configured. However, if a different MOH file is needed it would be configured here.

moh music-on-hold.au


STEP 11: OTHER CONFIGURATION
Other configuration to add for CME relating to call-forwarding and transferring calls within the
voice network are the following:

call-forward pattern .T
call-forward system redirecting-expanded
transfer-system full-consult dss
transfer-pattern 9.T
secondary-dialtone 9

STEP 12: DEFAULT (EXAMPLE FROM CISCO UC520)
A lot of defaults will be added under telephony-services not configured by the engineer, that
may include some of the following as an example.

max-ephones 14
max-dn 56
load 7914 S00104000100
load 7902 CP7902080001SCCP051117A
load 7906 SCCP11.8-0-3S
load 7911 SCCP11.8-0-3S
load 7921 CP7921G-1.0.1
load 7931 SCCP31.8-1-1SR2S
load 7936 cmterm_7936.3-3-5-0
load 7960-7940 P0030702T023
load 7941 TERM41.7-0-3-0S
load 7941GE TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S
load 7970 term70.default
load 7971 TERM70.7-0-3-0S
create cnf-files version-stamp 7960 Mar 10 2009 14:54:25


4.2.2 Directory Number (DN) Configuration (ephone-dn)

Once CME has been configured, the next part is to configure the DN or extensions that would
be used on the phones.

In our configuration example we will configure four DN that will later be mapped between two
IP Phones on our network.

Below we will configure a DN of 1001 (configured as number 1001). If this DN is associated
to one of the IP Phone buttons that label or display for that extension would read 1001
(Office). A username can be associated to this DN (configured as name 1001). If a caller
tries to call DN 1001 and there is no answer within 15 seconds or the line busy (because we
are on another call or the phone is not registered) the call is forwarded to 6000, which is our
DN for voicemail. Below shows that configuration:

ephone-dn 3 dual-line
number 1001
label 1001 (Office)
name 1001
call-forward busy 6000
call-forward noan 6000 timeout 15

We will configure two for DNs using extensions 1002 and 6700:

ring internal
number 1002
label 1002 (Family Room)
name 1002

For DN 6700, we will add a description that would display the external number right above
our lines/extensions on our IP phone:

number 6700
label 6700 (Main)
description 9252302203
name 6700

The next DN will be for extension 3001 and all call received will be forwarded to DN 4001
automatically:

ephone-dn 5
number 3001
call-forward all 4001


4.2.3 IP Phone Configuration (ephone)

Once the directory numbers have been configured on our CME router next we want to
configure our IP Phones and associate specific DNs to the phone. In this example, we will
use two Cisco 7970 IP phones.

NOTE: Make sure to configure DHCP before connecting IP Phones. You can
reference section 4.3 for how to configure DHCP on a Cisco IOS device.

For Phone1, that IP Phone and its MAC address will be listed under ephone 6. Its type will
automatically be provided to us once the phone starts up, so we dont need to configure that.

For button 1 on our IP phone it will use DN profile 1, which is for 6700.
For button 3 on our IP phone it will use DN profile 5, which is for 3001, but it is
configured to forward all calls to DN 4001.

ephone 6
mac-address 0011.932B.8B15
type 7970
button 1:10 2:3 3:5

For Phone2, that IP Phone and its MAC address will be listed under ephone 2. Its type will
automatically be provided to us once the phone starts up, so we dont need to configure that.

For button 1 on our IP phone it will use DN profile 1, which is for 6700. Making this a
shared line now used between both 7970 phones.

ephone 2
mac-address 001C.58F0.7619
type 7970
button 1:10 2:4

NOTE: If Auto-Registration is configured new phones plugged in will use the temporary
number hence that new phone will be added with a new ephone listed in our
configuration including its MAC address. We can simply just locate that ephone and
re-configure the buttons with its new extension.

4.3 Voice and Data VLAN Configuration

Two of the biggest best practices and recommended discussed with all IP Telephony
Solutions are with the following points:

1) Implement End-to-End QoS giving Voice RTP traffic high priority
2) Create a separate network (VLAN) for your Voice traffic

Here is the configuration for the second common point discussed.

1. First, on your L2 network configure two VLANs, one for Data and the other for Voice.

vlan 10
name RHG-VLAN-DATA

vlan 100
name RHG-VLAN-VOICE

2. Next we will configure a switch port that has a PC connected into an IP Phone, which
is then plugged into a switch port reflecting our DATA and VOICE VLAN assignment.
Let's assume this port is for Fa0/1.

interface FastEthernet0/1
description EDGE: VLAN DATA+VOICE
switchport mode access
switchport voice vlan 100
spanning-tree portfast

3. Let's say that this configuration is on a Access Switch connecting into a Core or
Aggregation switch configured for these two VLANs. Well this uplink port needs to
be configured for 802.1Q to carry our two VLAN tags across. We will also be specific
in our configuration and only allow our two configured VLANs. Let's assume this
uplink port is Gi0/1.

interface GigabitEthernet0/1
description UPLINK: LAN CORE OR AGG
switchport trunk allowed vlan 10,100
switchport mode trunk
switchport nonegotiate
spanning-tree portfast trunk

4. If the routing for our Data and Voice VLANs are configured on a L3-switch (likely our
LAN Core and Aggregation L3-switch in our network) then we can configure the SVI
interface for the two VLANs making them routable. Below is that configuration:

interface Vlan10
description SVI: VLAN DATA
ip address 10.67.79.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp

interface Vlan100
description SVI: VLAN VOICE
ip address 10.67.78.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp

Configuration Summary

vlan 10
name RHG-VLAN-DATA

vlan 100
name RHG-VLAN-VOICE

description EDGE: VLAN DATA+VOICE
switchport mode access
switchport voice vlan 100

interface GigabitEthernet0/1
description UPLINK: LAN CORE OR AGG
switchport trunk allowed vlan 10,100
switchport nonegotiate
spanning-tree portfast trunk

interface Vlan10
description SVI: VLAN DATA
ip address 10.67.79.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp

interface Vlan100
description SVI: VLAN VOICE
ip address 10.67.78.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp

4.4 Configuring DHCP on Cisco IOS

To enable DHCP services on our network this can be configured on a Cisco router, L3-
switch, or firewall, but its recommended to enable DHCP services on a server. But, for
small environments we can enable DHCP on our Cisco IOS router that is also running CME.

Below we will configure a DHCP scope for the 10.67.78.0 network assigning usable IP
addresses starting with 10.67.78.30 to 10.67.78.254. We will define the default gateway and
DNS servers the devices (our IP Phones) would use. We will also include option 150 with
the IP address of the CME router. This is important to specify the location of our CME router
or our connected IP Phones will not know how to register with the phone system. Option 150
points to the TFTP server, which happens to be our CallManager Express router. The TFTP
will supply info to the IP Phones for which Call Processing server to register with.

ip dhcp excluded-address 10.67.78.1 10.67.78.29

ip dhcp pool ROUTEHUB-DHCP-LAN-POOL
network 10.67.78.0 255.255.255.0
default-router 10.67.78.1
option 150 ip 10.67.78.1
dns-server 206.13.28.12 64.169.140.6
lease infinite

4.5 Mapping an analog line (DID) to an IP phone

This is a commonly question asked, but rare to find how it can be configured with CME. First,
lets explain what we are talking about. Lets say we have three IP Phones on our network
and we have three phone lines with a dedicated phone number (or DID number) associated
to each phone line. Well how can we configure each analog line & DID to be mapped to one
IP Phone internally for incoming and/or outgoing calling. Here is how we would configure
that on our CME router.

In our configuration example, we will assume we have two IP phones and two analog lines.
Phone1 will use extension 201 and Phone2 will use 202. Each phone will be tied to one of
the analog lines. For external calling Phone1 and Phone2 would need to dial 9 first then the
full number.

STEP 1: TRANSLATION RULES
First we will configure two translation rules for our two phones. For phone1, if it dials 9 first
then the full number like 1-925-230-2203 then we want to translate the 9 to 19. For
Phone2, another translation is configured where we would translate 9 to 29.

voice translation-rule 1
rule 1 /^9/ /19/

rule 1 /^9/ /29/


STEP 2: TRANSLATION PROFILE
Next we will associate each of the configured translation rules to their own profile for each
phone under that phones extension in the name. This translation would happen with the
number we would dial from our IP phone or the called number.

voice translation-profile TP-201
translate called 1

voice translation-profile TP-202
translate called 2

Lets quickly discuss calling and called. Calling is when we are specifying the source or
the caller in a phone conversation. Called would be our callee or destination number that
is dialed.

For example, a person at 800-123-4567 is calling someone at 925-230-2203. Our 800
number would be our calling or source and our 925-230-2203 would be our called or
destination.

STEP 3: DIAL PEER (VOIP)
Next we will configure two VOIP dial peers for each phone that will associate one of the
translation profiles configured. Plus we will include what the calling number or answer-
address for all calls matching the extension for each phone. So, for example, any number
that is dialed by phone1 at extension 201 would automatically match that dial peer configured
(in our example that would be VOIP dial peer 1). Matching to that dial peer would use the
translation rules and translate the access code of 9 to 19 if that matches.

dial-peer voice 1 voip
translation-profile incoming TP-201
answer-address 201

translation-profile incoming TP-202
answer-address 202

STEP 4: DIAL PEER (POTS)
Now we would configure two POTS dial peers mapped to an FXO port which has a dedicated
analog line plugged in. Each dial peer route pattern will have the new translated access code
(which can be 19 or 29). So, if phone1 was making an external code and its access code
was translated from 9 to 19 then it would use the configured POTS dial peer 19 routing the
call through that analog port. The T in the translation means any full phone number, but will
strip off the 19 or 29 before going out to the PSTN.

dial-peer voice 19 pots
destination-pattern 19T
port 1/0/0

destination-pattern 29T
port 1/0/1


STEP 5: VOICE PORT FOR INCOMING CALLS
Steps 1 to 4 was focused on configuring the phones for outgoing calls through there
dedicated analog line. Well what about incoming calls being routed to a specific IP phone. If
a caller calls the user at Phone1 using its external phone number we want that call to be
forwarded to Phone1. The configuration is straight-forward by using the PLAR command
followed by the extension we want all calls to be sent to among other necessary configuration
including enabling caller-ID on the FXO analog port.

voice-port 1/0/0
supervisory disconnect dualtone pre-connect
pre-dial-delay 0
no vad
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 201
caller-id enable

voice-port 1/0/1
pre-dial-delay 0
no vad
caller-id enable

4.6 Configuring FXS port as a SCCP port

By default when analog phones are plugged into analog modules like FXS ports they are not
part of CME or its features such as Hunt Groups. However, there is a way to configure a
FXS port that has a connected analog device like a phone to be a SCCP port meaning it can
be part of CME which uses SCCP for all communications with IP phones.

STEP 1: SCCP
First lets configure globally SCCP to be binded to the BVI10 interface that is being used for
CME and phone communication. The IP address on BVI10 is 10.67.78.1

sccp local BVI10
sccp ccm 10.67.78.1 identifier 1 priority 1 version 4.1
sccp

sccp ccm group 1
bind interface BVI10
associate ccm 1 priority 1
associate profile 1 register mtp001d4567c690
keepalive retries 5
switchback method graceful

For the line, mtp001d4567c690, the last part is the mac address from the BVI10 interface.

uc01tra#show interfaces bvI 10
BVI10 is up, line protocol is up
Hardware is BVI, address is 001d.4567.c690 (bia 001b.8faa.a860)

STEP 2: STCAPP
Next we will enable STCAPP and associate our configured SCCP group to this application.

stcapp ccm-group 1
stcapp

STEP 3: CONFIGURE FXS PORT AS A SCCP PORT
Next we will configure our FXS port (located on port 0/0/0 on our Cisco UC520 appliance or
router) to be an SCCP port to be able to communicate with CME. We will also enable caller-
ID on this port.

service stcapp
port 0/0/0

voice-port 0/0/0
caller-id enable


STEP 4: GET THE MAC ADDRESS FOR THE FXS PORT
Next, get the MAC address associated with port 0/0/0, which will look like this
...AN1D4567C690000.

This info can be obtained by issuing the following command:

uc01tra#show stcapp device summary
Total Devices: 1
Total Calls in Progress: 0
Total Call Legs in Use: 0

Port Device Device Call Dev Directory Dev
Identifier Name State State Type Number Cntl
---------- --------------- -------- ------------- ------- ----------
- ----
0/0/0 AN1D4567C690000 IS IDLE ALG 6776 CME

There we see AN1D4567C690000

STEP 5: ADD MAC ADDRESS UNDER A NEW EPHONE FOR CME
Next we will configure a new ephone for our FXS port with the MAC address we determined
from step 4. The type would be ANL and will use the directory number found at ephone-dn
10.

ephone 4
device-security-mode none
mac-address D456.7C69.0000
type anl
button 1:10

Now our FXS port can place/receive calls including access to other services like voicemail or
hunt groups on CME.

4.7 CME as SIP Server for SIP Clients

Cisco CME router can also be configured to be a SIP server to accept SIP phones on the
network working with CME.

STEP 1: CONFIGURE SIP SERVER
First, lets enable our router as a SIP server globally. Most are defaults and our SIP server
will use the IP address from the router itself listening on the default SIP port number, 5060.
Our SIP Server will accept up to 12 directory numbers (or extensions) and 12 phone devices.

voice register global
mode cme
source-address 10.67.78.1 port 5060
max-dn 12
max-pool 12
timezone 47
time-format 24
date-format YY-M-D
dst start Oct week 8 day Sun time 02:00
dst stop Mar week 8 day Sun time 02:00

STEP 2: DIRECTORY NUMBERS
Next we need to configure our directory numbers that would be used for our SIP clients. In
our configuration we will configure two directory numbers, 8701 and 8778.

voice register dn 1
number 8701
name ROUTEHUB SIP client (X-lite)

voice register dn 2
number 8778
name Michel Thomatis (SIP)

STEP 3: DEVICE AND DN ASSOCIATION
Now we will configure our SIP phone profile to specify the ID or mac address of the SIP
client, the directory number it will use configured from step 2, the codec, and the
username/password that the SIP client will use to authenticate with the SIP server.

voice register pool 1
id mac 000C.F179.1682
number 1 dn 1
username 8701 password cisco6778
codec g711ulaw

id mac 0019.D111.D2E8
number 1 dn 2
codec g711ulaw

NOTE: Unless this is already configured it is best to configure our CME router to
accept SIP connections with the following configuration among other allowed
protocol communications with H.323.

voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12

STEP 4: USING THE SIP CLIENT
Once the service has been configured, download and install a SIP client (like X-lite) on a
computer. Under the SIP profile settings specify the IP address of the SIP server (our CME
router), our DN, username, and password. With everything setup the SIP client would be
register to the SIP server router ready to place and receive calls.

4.8 Blocking incoming calls from PSTN

This is a common request for knowing how to block certain phone numbers of calls received
from the PSTN like many telemarketers. Well here is the configuration to block certain
numbers.

In our example, we are blocking 800 number 800-123-4567.

STEP 1: VOICE TRANSLATION RULE
First we need to configure a translation rule to match and reject the 800 number in question
we want to block:

rule 1 reject /8001234567/

If we want to block other numbers then those would be added the same as rule 1, but would
be added as rule 2 and so forth.

STEP 2: VOICE TRANSLATION PROFILE
Next, we will create a voice translation profile called call_block that will be map our
configured translation rule from Step 1.

Here we are specifying the source or the caller in a phone conversation. This would be
calling. Where called would be our callee or destination number that is dialed.

For example, a person at 800-123-4567 is calling someone at 925-230-2203. Our 800
number would be our calling or source and our 925-230-2203 would be our called or
destination. We want to block numbers from certain calling or source numbers into our
voice network.

This is what our profile configuration will look like:

voice translation-profile call_block
translate calling 5

STEP 3: DIAL PEER
Next we will configure a voice POTS dial peer. In our environment we have a single analog
line from our PSTN with a single DID number. This analog line is plugged into port 0 on our
FXO module. All calls placed and received are going through this single analog line.

We will apply our call_block translation profile configured from step 2 to our dial peer that is
associated to port 0/1/0 (which is our FXO port connected to our PSTN). This is the same
dial peer that is used for placing calls when users internally dial a 9 first then the full number
(which is represented as T) for anything local or long distance. All incoming calls coming
into our voice router will match this dial peer because of the syntax incoming called-number
.

call-block translation-profile incoming call_block
call-block disconnect-cause incoming call-reject
destination-pattern 9.T
incoming called-number .
port 0/1/0

When that 800 number calls into our voice network it will match this dial peer we configured
and the caller will hear a fast busy because the voice router will reject and disconnect the
call.

4.9 Setting up an Authenticated SIP Trunk to SIP Provider

Cisco CME router can be configured to form a single authenticated SIP trunk to another SIP
component such as another SIP router, server, or even a SIP Provider like ViaTalk. A SIP
trunk can be authenticated (meaning it requires a username and password to be established)
or a non-authenticated SIP trunk (no authentication needed and many SIP trunks can be
established). Special call routing, patterns, and translations are needed for placing/receiving
calls through the SIP provider especially if a PSTN provider is connected to the CME router
for external calling.

STEP 1: AUTHENTICATED SIP CONFIGURATION
First we need to configure our authenticated SIP trunk to a SIP provider like ViaTalk. In our
case, the username would be our dedicated number and the password would be provided to
us through the providers control panel. We also need to specify the SIP providers server IP
address or host/domain name.

sip-ua
authentication username 19252302203 password cisco6778
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sanfrancisco-1.vtnoc.net expires 3600
sip-server dns:sanfrancisco-1.vtnoc.net
host-registrar

STEP 2: ALLOW SIP CONNECTIONS
Next we need to allow SIP connections on our router with other devices that may be SIP or
H.323 connections among other details.

voice service voip
allow-connections h323 to h323
sip
registrar server expires max 3600 min 3600
localhost dns:sanfrancisco-1.vtnoc.net

STEP 3: DIRECTORY NUMBER FOR SIP NUMBER
Next we will configure a new ephone directory number that will use extension 7700 and its
full DID number would be 925-230-2203, our dedicated SIP number.

number 7700 secondary 19252302203
name 7700


STEP 4: TRANSLATION RULE AND PROFILE
Next, we need to configure two translation rules. In our voice network if we dial 8 first then
the full number it will be routed across our SIP trunk. If we dial 9 first then the full number
that would be routed across our PSTN/analog connection.

In our first translation rules any digits that match 8 first then a series of numbers it will strip off
the first digit, which is our access code of 8. Leaving us with the full number that would be
routed across the SIP trunk. This translation would be applied for calls that we make from
the inside.

rule 1 /^8(.*)/ /\1/
rule 2 /^8$1[2-9].........$$/ /\1/

voice translation-profile routehub-tp-sip-outgoing
translate called 2

In our second translation rule any digits or directory number it sees as the source (like 7700,
which we configured) will be translated to its full DID number of 925-230-2203. This means
we can place calls across our SIP trunk from other DNs other than 7700.

rule 1 /^.*/ /19252302203/

translate calling 3

STEP 5: VOIP DIAL PEER
Last we will configure our dial peer required for call routing. This will associate the translation
profile that we configured from step 4 and contain a route pattern with 8 first (our access
code) followed by a pattern numbers that match any local or long distance call. This would
be routed across our authenticated SIP trunk (since we specified session target sip-server).
We will also include codecs and DTMF details.

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8

description **Outgoing Call to SIP Trunk**
translation-profile outgoing routehub-tp-sip-outgoing
destination-pattern 8[0-1][2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad

STEP 6: PLACING CALLS ACROSS SIP TRUNK
To place calls across our SIP we would dial 8 first (our access code) then the full number.
That would match dial-peer 12 that would be routed to our SIP provider. All other calls
through our PSTN require access code 9 that would be routed through our FXO port.

STEP 7: PHONE REGISTRATION WITH SIP UA
The one thing that occurs with a SIP UA configuration like in our example is that all directory
numbers configured will try to register itself with the authenticated SIP trunk. We only want
our DN of 7700 to be registered with the SIP trunk. Thus, under each DN (ephone-dn) we
will add the command no-reg primary to NOT register with the SIP trunk as a best practice.

In our configuration we will do this for DNs 1001, 1002, and 6700:

number 1001 no-reg primary



4.10 Phone Directory

You can configure a Phone Directory on our CME router to include a local directory with a list
of names pre-configured to be accessed directly from any phone registered with that CME
router by pressing the Directory button on the phone then choosing Local Directory.

Below is how we can configure names and numbers (extensions or full DID numbers) on
CME:

telephony-service
directory first-name-first
directory entry 1 919252302203 name ROUTEHUB (Main)
directory entry 2 912098329950 name Deliver Ease (Main)
directory entry 3 912091234567 name Misc Number (Cell)

If the number in one of the entries is external, then the access code (like 9 in our
configuration) needs to be included.

Also, any other directory added to CME (SIP or SCCP) will also be included in our local
directory listing.

As a recap to access the phone directory press the Directory button on the phone then
choose Local Directory where we can search based on First name, Last name, and/or
number. Or we can simply press Submit with nothing inputted to display all entries in the
directory.

4.11 Single Number Reach (SNR)

Single Number Reach (SNR) is a feature that is available with CME version 7 and higher.
SNR is a feature that is branded as no more phone tag, which means any person who calls
a certain extension enabled for SNR, CME can call another phone number at the same time.

In this configuration, if we dial 1002 (or it's full DID number) it will automatically ring 1001. If
there is no answer then the call would go to voicemail (at 6000)

mobility
snr 1001 delay 5 timeout 15 cfwd-noan 6000

Or our SNR number may be an external number like a cell phone or some other external
number. However, in our environment to place outgoing calls we need to dial 8 first. This is
reflected in our configuration below. Our external number would be our main number for
ROUTEHUB.


4.12 Sending Calls to Voicemail (CUE)

This configuration on CME is needed for sending calls to a voicemail system via a SIP
connection particularly to a CUE service engine. But, we want to show you the necessary
configuration needed for calls to be sent to voicemail (to Cisco Unity Express) if there is no
answer to a particular directory number.

STEP 1: DIAL PEER CONFIGURATION
First we need to configure a dial peer to match our voicemail pilot number (this is the number
where we want to send calls to voicemail and/or to access our voicemails). In our
configuration that would be DN 6000. This directory number would be our destination pattern
that would be forwarded to the CUE service engine found at IP address 192.168.5.2, which
will be a SIP trunk connection.

destination-pattern 6000
session target ipv4:192.168.5.2
dtmf-relay sip-notify
codec g711ulaw
no vad

STEP 2: VOICEMAIL BUTTON CONFIGURATION ON CME
Next we will add our voicemail pilot number of 6000 under our CME telephony service. This
configuration will setup a direct speed dial to access our voicemail. When we press the
Mail button on our phone it will dial the voicemail pilot number.

telephony-service
voicemail 6000

STEP 3: SENDING CALLS TO VOICEMAIL ON CME
When the line (or directory number) is busy or is not answered the call would be forwarded to
voicemail. In our configuration if someone is calling 6700, but the line is busy or there is no
answer (within 15 seconds) then the call would be forwarded to voicemail at directory number
6000, which will match the dial peer we configured in step 1.


STEP 4: CONFIGURE MWI
Next we need to configure our Message Waiting Indictor (MWI). This means that if a new
voicemail message has arrived, CUE will send a MWI ON message to the number where the
message was left. A red light would turn on the phone. Once the voicemail message is read
and no longer new then a MWI OFF message would be sent to turn off the red light.

In our configuration, our MWI ON directory number will be 8000 and our MWI OFF directory
number will be 8001. You will also notice that the MWI directory number include . (four
dots) which represents the directory number that is receiving the MWI message. So, if a
voicemail message is left for 6700 then the following MWI message is sent: 80006700. Once
the voicemail message is heard and no longer new then the following MWI is sent:
80016700.

ephone-dn 20
number 8000.... no-reg primary
mwi on

ephone-dn 21
mwi off

4.13 Conferencing

Conferencing is a feature that allows multiple participants or callers to join a single call.
Conferencing with a feature called MeetMe allows you to setup conference bridges using a
MeetMe directory number on the CME router and allow callers to call directly into the
conference bridge.

Ad-hoc conferencing is setup by the user from their IP phone by adding another person to an
existing call. No one can dial into an Ad-hoc conference directly, so its important to
understand when and how to use these conferencing services on the network.

4.13.1 MeetMe Conferencing

STEP 1: SCCP CONFIGURATION
This may have been configured already if a feature like enabling an FXS port as a SCCP port
has been configured.

If not configure a SCCP (Cisco Skinny) profile globally binded to the BVI10 interface that is
being used for CME and phone communication. The IP address on BVI10 is 10.67.78.1

sccp local BVI10
sccp

sccp ccm group 1
keepalive retries 5

For the line, mtp001d4567c690, the last part is the mac address from the BVI10 interface.

uc01tra#show interfaces bvI 10
BVI10 is up, line protocol is up
Hardware is BVI, address is 001d.4567.c690 (bia 001b.8faa.a860)

STEP 2: VOICE CLASS CUSTOM TONES
Next, configure the tones and frequencies for when callers join a call or leave a call for a
conference bridge.

voice class custom-cptone routehub-leave
dualtone conference
frequency 900 900
cadence 150 50 150 50

voice class custom-cptone routehub-join
dualtone conference
frequency 1200 1200
cadence 150 50 150 50


STEP 3: DSPFARM CONFIGURATION
Within our voice-card on our CME router we need to enable dsp resources (dspfarm). DSP
resources are critical for many voice interfaces (like PRI lines) and services like conferencing
within a Cisco IOS router.

Our DSPFARM for conferencing will include our configured custom tones, the use of the
SCCP application (which is required in order for our conferencing service to interact with
CME), the number of conference sessions (in our configuration that would be two), and list a
supported codecs on the system.

dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
conference-join custom-cptone routehub-join
conference-leave custom-cptone routehub-leave
associate application SCCP

STEP 4: ENABLE CONFERENCING UNDER TELEPHONY SERVICES
Next we will enable conferencing and our dspfarm profile under our CME service specifying
that we are using a hardware-based conferencing solution via the DSPFARM configuration
we recently completed.

telephony-service
max-conferences 8 gain -6
sdspfarm conference mute-on 11 mute-off 12
sdspfarm units 3
sdspfarm tag 1 mtp001d4567c690
conference hardware

STEP 5: CONFIGURE MEETME DIRECTORY NUMBERS
Next will configure our MeetMe directory numbers to be 6999 and enable conference
meetme across four ephone-dn profiles for load distribution.

number 6999
conference meetme
no huntstop

number 6999
conference meetme
preference 1
no huntstop

number 6999
conference meetme
preference 2
no huntstop

number 6999
conference meetme
preference 3
no huntstop

STEP 6: USING THE CONFERENCE BRIDGE (WITH MEEME)
The first thing that needs to happen is the conference bridge needs to be setup first by a
moderator or any person who is the lead during a group call. We do this by lifting the phone
off the hook (any phone within the voice network is fine) then locate the MeetMe softkey on
the phone.

NOTE: If no MeetMe softkey exist then follow the section within this document
discussing how to setup a Phone Softkey template.

Pressing the MeetMe softkey you will hear two beeps. There we would input our MeetMe
number of 6999. Moments later the conference bridge will be setup and the moderator
person would be the first person to join the conference call, which they will see on their
phone as Conference.

Now all other callers can now dial 6999 or the full DID number mapping to 6999 directly to
join the conference call.

From there any person including the moderator can leave the conference call at any time by
simply hanging up the call.

Another great thing you can do is to view the callers on the conference call by pressing the
ConfList softkey. From that same list you can also remove any participant from the call as
needed.

4.13.2 Ad-Hoc Conferencing

The configuration for Ad-Hoc conferencing is similar to the MeetMe conferencing steps 1
through 4. Step 5, for our DN numbers will be a little different.

STEP 1: CONFIGURE ADHOC DIRECTORY NUMBERS
We will configure our Adhoc directory number to be 6998 and we will enable conference ad-
hoc across two ephone-dn profiles for load distribution.

number 6998
name Conference
conference ad-hoc
preference 1
no huntstop

number 6998
name Conference
conference ad-hoc
preference 2
no huntstop

STEP 2: USING THE CONFERENCE BRIDGE (WITH AD-HOC)
Ad-hoc conferencing is a little different to use compared to MeetMe conferencing. In Ad-hoc
our conference bridge is built and controlled by the user. No caller can call into an Ad-hoc
conference bridge, they can only be joined into an existing call by the user.

Lets say we have a call setup with someone on the outside. But we want to bring another
person into the call. First, we would locate and press the Confrn softkey which will place a
new call. This is where we call the second caller we want to join into the existing call. The
first caller will be placed on-hold hearing hold music until the call is completed.

When the call is connected, we can simply press the Confrn softkey again and now we will
have three callers on the same call.

NOTE: If no Confrn softkey exist then follow the section within this document
discussing how to setup a Phone Softkey template.

4.14 Paging

Intercom and Paging are two features that work with CME and really operate the same way,
but with different results. Paging is a feature that allows you to do announcements or
broadcasts to groups of phones or all phones on the network with only one-way
communication. Meaning if I setup a Page with a group of people and I speak they cannot
communicate back. This would be good for emergency broadcast messages.

STEP 1: CONFIGURE PAGING
First we need to configure Paging and assign what the directory number will be. This can
also be configured or enabled for multicast.

In our example, we will configure Paging to use directory number 6001 plus it will use
multicast address 239.192.2.1 with port number 2000.

ephone-dn 1
number 6001
name ROUTEHUB Paging System
paging ip 239.192.2.1 port 2000

STEP 2: ENABLE PAGING UNDER EACH PHONE (EPHONE)
Next we will enable paging for each phone with the profile and directory number we
configured from step 1.

ephone 2
paging-dn 1

ephone 5
paging-dn 1

STEP 3: USING THE PAGING SYSTEM
To use the paging system from any phone we would dial the DN for paging, which would be
6001. This would do a broadcast with to all phone that were associated with that paging
profile (paging-dn 1). This would be ephone 2 and ephone 5. Remember this is a one way
communication from one to many.

4.15 Personal Speed Dial

To setup a personal speed dial or fast dial for a users phone go to that physical ephone
device on the CME router then add that fast dial including the directory number (or DID
number) including the label/name for that number.

Here we will configure a fastdial for number 1001 with a description or label of FR1002 under
our IP phone (configured under ephone 6).

ephone 6
fastdial 1 1002 name FR1002

To access the Personal Speed Dials go to the Directories button on the phone directly then
go to Personal Speed Dials to access your list of fast dials.

4.16 Upgrading CallManager Express

The best and fastest way to upgrade the Cisco CallManager Express (CME) software version
is upgrade the router (or UC520) IOS itself.

For example, on the Cisco UC520, I was at CME version 4.2

I upgraded to: uc500-advipservicesk9-mz.124-22.YB.bin

Completing the upgrade provided me with CME version 7.1 installed and ready to use with
new features such as Single Number Reach (SNR), which is available within this document.

4.17 Intercom

Intercom and Paging are two features that work with CME and really operate the same way,
but with different results. Intercom is a feature that allows you to do announcements or
broadcasts to groups of phones on the network with two-way communication. Meaning if I
setup an Intercom with a group of people and I speak then they can communicate back.

STEP 1: CONFIGURE INTERCOM DIRECTORY NUMBERS
First we need to configure two directory numbers for each of the two-way participants.
Configuring Intercom directory numbers looks different and includes an A at the beginning.

In our configuration our first Intercom DN will be 5001 (or A5001) that can initiate an Intercom
connection with the second Intercom DN 5002 (or A5002). And vice versa where the second
Intercom DN of 5002 can initiate an Intercom connection with the Intercom DN 5001.

ephone-dn 11
number A5001 no-reg primary
label Intercom
name Intercom
intercom A5002

ephone-dn 12
label Intercom
name Intercom
intercom A5001

STEP 2: ASSOCIATE INTERCOM DN WITH PHONES.
Next we will associate the two Intercom DN configured from step 1. One phone will be
associated with Intercom DN 5001 on button #2 for one Cisco 7970 phone. The second
phone will be associated with Intercom DN 5002 on button #2 for another Cisco 7970 phone.

ephone 6
type 7970
button 1:10 2:11 3:13 4:3

ephone 2
type 7970
button 1:10 2:12 3:4

STEP 3: USING INTERCOM
Now everything is configured. To start an Intercom connection from either IP phone simply
press the Intercom line on the phone then we will automatically connect to the other mapped
Intercom line configured. The call will automatically be answered and doesnt require
someone to answer the Intercom call. Remember its a like broadcast type service.


4.18 Hunt Group

A Hunt Group is a feature where a single number (or pilot number) is associated with one or
more dedicated phones/extensions. For example, when someone calls support it would be
the pilot number. On the backend, it will call each participant in the hunt group one-by-one
until the call is answered. If the call is not answered in a specific amount of time the call may
be routed to voicemail or some other number.

In our configuration example, we will configure our hunt group pilot number to be 6701. The
two directory numbers associated with this hunt group will be 6776 and 6700. If there is no
answer within 15 seconds the call is routed to voicemail at 6000.

ephone-hunt 1 sequential
pilot 6701
list 6776, 6700
final 6000
preference 1
timeout 15, 15

The hunt group works by a caller dialing 6701 which would proxy in a way to the first DN in
the configuration (if it is not busy), 6776 then to 6700 after 15 seconds.

4.19 Call Park

Call Park is a feature that allows a user to answer a call, place it on PARK (or HOLD really)
then retrieve the call from another phone on the network by inputting the Call Park directory
number.

Call Park is configured under the ephone-dn, and in our example we will configure Call Park
to use directory number 6002 with a timeout of 30 seconds. Meaning we have 30 seconds
to retrieve the call once it has been parked.

ephone-dn 2
number 6002
park-slot timeout 30 limit 10
name ROUTEHUB CALL PARK

To use call park, if we receive a call and we want to transfer the call to another phone within
the network, but we dont know which one. First we would locate the Park softkey and
press it. This will place the caller on-hold and a 30 second timer starts that we need to
retrieve from another phone. On the phone we will see the Call Park number of 6002
displayed.

We would go to another phone and dial the call park number of 6002 to retrieve the call.


4.20 How to setup Phone Softkey templates

If you want to change the layout of the softkey buttons on your phone or add additional
softkey buttons for additional services like conferencing or Live record then you need to
create a template with what the softkey layout will look like. That template would then be
added to the phone(s) and require a reset.

STEP 1: PHONE TEMPLATE
First, create the phone template you want in terms of the softkeys including the layout of the
softkeys that would be displayed. The softkey buttons will vary depending on what we are
doing with our phone. For example, the template will differ depending whether a call is
placed on-hold, the phone idle, or if we have a connected call.

Below we will configure our phone template.

ephone-template 1
softkeys hold Newcall Resume Select Join
softkeys idle Redial Newcall Cfwdall Pickup ConfList Dnd
softkeys seized Redial Pickup Meetme Endcall
softkeys connected Endcall ConfList Confrn Hold Join Park RmLstC

So if we have a connected call with someone, from our phone we will see End Call,
Conference List, Conference, Hold and other softkey buttons available that we can do
with that active call based on the template we setup.

STEP 2: ASSOCIATE PHONE TEMPLATE
Next step is to associate the phone template under our ephone profile. In our configuration
we will apply this template under one of our Cisco 7970 phones.

ephone 6
ephone-template 1
type 7970

Once that is done we will need to reset our phone to make our new template become ready
to use:
ephone 6
reset

4.21 Extension Mobility

Extension Mobility is a feature, very popular with the Cisco Unified Communication Series,
that allows you to login to any IP Phone on the network and your phone profile (consisting of
your number and speed dials) will be loaded on that phone. This is done by creating a profile
and is configured with your directory number and other info. This feature allows users to not
be physically restrained to their phone or location.

STEP 1: EXTENSION MOBILITY PROFILE
First we need to configure our Extension Mobility profile. We will configure two profiles for
two phones on the network. They will contain the directory numbers on that phone today
including the username/password for authentication.

voice user-profile 1
pin 6778
user 78 password 78
number 6700,A5001,7700,1001 type feature-ring

pin 6700
user 70 password 70
number 6700,A5002,1002 type feature-ring

This configuration shows two phone profiles. One phone contains directory numbers 6700,
7700, and 1001. Another phone within our network has directory numbers 6700 and 1002.

STEP 2: LOGOUT PROFILES
The logout profile is basically identical to what we configured for our profile, but requires a
different username/password. The logout profile is what happens when a user logs out of
Extension Mobility and the current physical phone resets back to its default.

For example, in our configuration lets say we have two IP phones. One phone (user1) has
three directory numbers (6700, 7700, and 1001) and our second IP phone (user2) has two
lines (6700 and 1002). Two profiles were created globally on the CME router. Lets say
user1 is at user2s desk. User1 can login to user2s phone to use their own directory
numbers of 6700, 7700, or 1001. When user1 is finished they would need to logout from
user2 phone, so the phone can be useable for user2. This is where the logout profile comes
into play, which is to restore the physical phone back to its default configuration.

Below we will configure our two logout profiles, but with a different username/password:

voice logout-profile 1
pin 6778
user 16778 password 6778

pin 6700
number 6700,A5002,1002 type feature-ring


STEP 3: ASSOCIATE LOGOUT-PROFILE TO PHYSICAL PHONE (EPHONE)
Next under each phone (configured under ephone) we will place the logout profile matching
what that current phone has in terms of its directory numbers.

ephone 2
logout-profile 2

ephone 6
logout-profile 1

STEP 4: EXTENSION MOBILITY URL
Last we need to configure the phone service that would be used for users to login to the
Extension Mobility service that will upload our phone profile.

telephony-service
url authentication http://10.67.78.1/voiceview/authentication/authenticate.do

NOTE: the IP address is the IP on our CME router that is associated for our CME
configured initially under our telephony-service.

STEP 5: USING EXTENSION MOBILITY
The question now is how do you access the Extension Mobility service to login at any phone
enabled for extension mobility and have our phone profile uploaded.

To login, first, click on the Services button on the phone. Next go to the Extension Mobility
service listed, login (using the username/password configured under the logout-profile), and
the phone will reload with our authenticated profile. To logout, go back to the Extension
Mobility and click Logout.

4.22 How to setup a custom ring tone?

To create a custom ring tone for your IP Phone complete the following steps.

STEP1: CREATE PCM/RAW AUDIO FILES
First, choose the audio you want such as an mp3 file. This file needs to be converted to a
.raw format.

I use the program called Switch by NCH to convert a short playing mp3 file into a raw file
using 8khz, Mono.

In our example, we have an mp3 file called 24.mp3. This file would be converted to 24.raw
using our NCH Switch application.

STEP 2: CREATE THE XML FILES
Next we need to create two XML files: RingList.xml and DistinctiveRingList.xml

The most common XML file created is the RingList.xml, which will provide a ring tone for ALL
lines on a phone. Where the DistrictiveRingList.xml file allows a unique ring tone different for
certain lines. For example, maybe our Support line will have a unique or distinctive ring and
all other lines on our phone can have use a different ring tone.

This can be created using a program like Notepad on your Windows computer. Here is what
our XML files would like:

<CiscoIPPhoneRingList>
<Ring>
<DisplayName>24</DisplayName>
<FileName>24.raw</FileName>
</Ring>
</CiscoIPPhoneRingList>

In our example, we have listed one ring tone that shows the display name (what we will see
on our IP Phone) and the actual file name that we converted to a raw format that we will soon
upload to our Cisco router running CME. Our converted 24.raw file would be listed in our
XML file as an available ring tone.

STEP 3: UPLOAD XML AND RAW FILES
Next, we need to upload the XML files and the RAW files to the flash on the router.

Below, for example, we have a computer running a TFTP program. Its IP address is
10.67.78.243. We will upload the XML files we created from step 2 and our 24.raw converted
file into our flash. We would do the following one-by-one until each file is successfully
uploaded to our Cisco CME router.

copy tftp://10.67.78.243/24.raw flash:
copy tftp://10.67.78.243/ RingList.xml flash:
copy tftp://10.67.78.243/ DistinctiveRingList.xml flash:


STEP 4: TFTP CONFIGURATION
Our Cisco CME router is also configured as a TFTP server. The TFTP server is where all of
our phone loads and XML files exist.

With the files we recently uploaded we need to enable those files we be active and usable on
our voice network. Doing this is important because this is where our phones will look for any
available ring tones.

We will add these files into our TFTP server:

tftp-server flash:RingList.xml
tftp-server flash:DistinctiveRingList.xml
tftp-server flash:24.raw

STEP 5: CONFIGURE DISTINCTIVE RING
If we want to use a distinctive ring for a certain directory number then we need to configure
our distinctive ring by going into the ephone-dn for the directory number we want to be
unique and configure ring external. In our example, we want a distinctive ring for number
6700.

ring external

STEP 6: RESTART
Next either restart each phone one-by-one or all phones under the telephony-service.

If we have an IP phone configured under ephone 6 we would reset it here:

ephone 6
reset

If we want to reset all phones, so they could use the new ring tone, we would go under the
telephony-service to reset all registered devices.

telephony-service
reset

STEP 7: CHANGE RING TONE
Now we can change our ring tone to our custom tone configured.

Go to "Settings" > "User Preferences" > "Rings" > then select the ring tone for ALL lines
(reflected in our RingList.xml) or ring tone for each line (reflected in our
DistinctiveRingList.xml).

NOTES: It is recommended to use actual ring tone files (raw files) for your ring tones
over converting mp3 files to raw files. There are a lot of issues to consider. If the file
is too large it may not load or play especially with Cisco 7960 IP phones.

For troubleshooting tips:
Erase phone configuration. You do this by going to Services > and pressing **#
which will make the "Erase" softkey appear then you can delete the configuration
and the phone will reset.
Removing the TFTP server commands and resetting the IP Phone. Reapplying
the TFTP server commands and restart phones again.

I strongly recommend using actual ring tone files or the ring tones available on the
system. We spent hours realizing the issues with custom ring tones and realizing it's
not very important or a must have feature to use.

4.23 Call Center

Call Center is a feature that uses TCL scripts to provide a call center solution with CME. A
call center is like a Hunt Group or Shared line, but allows calls to go into a queue until an
available agent is ready and it allows agents to login and logout as needed.

STEP 1: CREATE DN FOR AGENTS
First configure two DN numbers that would be used for the agents in the call center
deployment with CME. In our configuration our agent DNs will be 2001 and 2002.

number 2001

number 2002

STEP 2: HUNT GROUP
Next we will configure a hunt group that will include the two DN we configured from step 1.
The pilot number will be 6721 and the timeout for ringing between each number listed in the
hunt group will be 10 seconds. We will also collect statistics for all call activity within this hunt
group.

ephone-hunt 1 longest-idle
pilot 6721
list 2001, 2002
timeout 10, 10
statistics collect

STEP 3: CALL CENTER STATISTICS
Under the telephony-service we will enable the location (TFTP server) where our hunt group
statistics (or reports) would be sent.

In our configuration, our TFTP server IP will be 10.67.78.243 placed under the DATA folder
and reports would sent to the TFTP server every 2 hours.

telephony-service
hunt-group report url prefix tftp://10.67.78.243/data
hunt-group report url suffix 0 to 200
hunt-group report every 2 hours

STEP 4: DOWNLOAD CALL CENTER/AA TCL SCRIPT
First we need to download the following two TCL script applications from cisco.com.

app-b-acd-aa-2.1.2.3.tcl
app-b-acd-2.1.2.3.tcl

Once you have downloaded these two TCL scripts we need to upload them to the FLASH on
our CME router. Here is an example on how we would do this for one of the files.

uc01tra#copy tftp flash:
Address or name of remote host []? 10.67.78.3
Source filename []? app-b-acd-aa-2.1.2.3.tcl
Destination filename [app-b-acd-aa-2.1.2.3.tcl]

That will ask for the TFTP server IP (which should be some computer installed with a TFTP
server application) and the file that will be copied. This would be app-b-acd-aa-2.1.2.3.tcl for
one transfer then app-b-acd-2.1.2.3.tcl for the second transfer.

Make sure both files are copied to the routers flash successfully before continuing.

STEP 5: CONFIGURE CALL CENTER APPLICATION & SERVICE
Once the two TCL script applications have been copied to the flash disk we will enable the
TCL script applications.

There is a lot of configuration and most are defaults, but we will explain the configuration
important to change to use with your environment.

In our configuration our Call Center pilot number (like hunt groups) will be 6720 and our
configured hunt group from step 2 is added to this call center solution as option 2 because of
the 2 in aa-hunt2. Also since we are adding one hunt group that would be configured as well
(listed as param number-of-hunt-grps 1).

We will enable queuing (hence the command param service-name queue) where callers
who call into 6720 and all agents are busy will be placed into the queue (supporting up to 5
callers at one time).

Callers in the queue will be in the queue for 300 seconds until it dials the call center pilot
number again. However, we have configured our retry time to be 2 so after another 300
seconds the call would be forwarded to voicemail, at DN 6000.

application
service aa flash:app-b-acd-aa-2.1.2.3.tcl
param aa-hunt2 6721
paramspace english index 1
param number-of-hunt-grps 1
param queue-len 5
param handoff-string aa
param dial-by-extension-option 1
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 6720
paramspace english location flash:
param second-greeting-time 30
param queue-manager-debugs 1

param call-retry-timer 15
param voice-mail 6000
param max-time-call-retry 300
param service-name queue

service queue flash:app-b-acd-2.1.2.3.tcl
param queue-len 5
param aa-hunt2 6721

STEP 6: DIAL PEER ASSOCIATED WITH CALL CENTER
Last, we will enable our call center application under a new dial peer matching the DN of our
call center pilot number 6720. This would then be routed back to itself (10.67.78.1, the IP of
our CME router) where this application is configured at.

service aa
incoming called-number 6720
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad

4.24 Fax to Email using T.37

This is a great feature on the UC520 series with the most recent CME version that allows you
to accept faxes (even on the same analog line you receive calls) and convert them to a TIFF
format, which is sent to an email address using a standard called T.37. This feature uses
TCL scripts (which is really an application) to use this feature.

STEP 1: DOWNLOAD TCL SCRIPTS
First we need to download the following two TCL script applications from cisco.com.

app_faxmail_onramp.2.0.1.3.tcl
app_fax_detect.2.1.2.2.tcl

Once we have downloaded these two TCL scripts we need to upload them to the FLASH on
our CME router. Here is an example on how we would do this for one of the files.

uc01tra#copy tftp flash:
Address or name of remote host []? 10.67.78.3
Source filename []? app_faxmail_onramp.2.0.1.3.tcl
Destination filename [app_faxmail_onramp.2.0.1.3.tcl]

That will ask for the TFTP server IP (which should be some computer installed with a TFTP
server application) and the file that will be copied. This would be
app_faxmail_onramp.2.0.1.3.tcl for one transfer then app_fax_detect.2.1.2.2.tcl for the
second transfer.

Make sure both files are copied to the routers flash successfully before continuing.

STEP 2: ENABLE TCP SCRIPT APPLICATIONS
Next, once the TCP scripts have been copied to the routers flash we need to enable these
applications:

application
service onramp flash:app_faxmail_onramp.2.0.1.3.tcl

service fax_detect flash:app_fax_detect.2.1.2.2.tcl
param fax-dtmf 2
param mode listen-first
param voice-dtmf 1


STEP 3: CONFIGURE FAX TO EMAIL USING T.37
Once the TCL applications has been installed then we can configure FAX to Email using
T.37. Here we would specify the IP address of our SMTP server where our FAX messages
would be sent via email once it is converted to a TIFF format.

We will specify some of the mail settings such as the subject line of the email message and
other similar information:

fax interface-type fax-mail
mta send server 10.67.78.6 port 25
mta send subject You Received a Fax!
mta send with-subject both
mta send postmaster sales@routehub.com
mta send mail-from hostname routehub.com
mta send mail-from username IncomingFax
mta send return-receipt-to hostname routehub.com
mta send return-receipt-to username ROUTEHUB
mta receive aliases routehub.com
mta receive aliases 10.67.78.6
mta receive maximum-recipients 10
mta receive generate permanent-error

STEP 4: CONFIGURE VOICE PORTS
Next we will configure our FXO voice-port (FXO supported today for this feature) connecting
to a PSTN to forward all calls to 6700.

voice-port 0/1/0
pre-dial-delay 0
no vad
caller-id enable


STEP 5: FAX DETECT
Our PSTN analog line plugged into FXO port 0/1/0 is also used by our office for placing and
receiving calls. How will it know whether its a fax call or a regular voice call? This happens
with the fax_detect script application we installed.

When a call (fax or voice) reaches our FXO port it will be forwarded to 6700, which will match
the below dial peer because of the line incoming called-number 6700 command. On this
dial peer will enable it for fax detect where it will listen for any fax tones. If there is a fax
tone received then that call is routed to a different dial peer, configured in the next step. If
there are no fax tones then the call is routed normally to CME and to the phone(s) using
directory number 6700 will ring.

service fax_detect
direct-inward-dial
port 0/1/0

Caution: One serious note about this operation for consideration since this has
been a reported concern from our clients. When callers call a number where the port
is enabled for fax detect this is what the experience will be. Lets say we dial the
DID number that is used with 6700. We would hear a single ring then a pause for
about 5 seconds or less. No other ringing. This is what the script is doing, listening
for any fax tones to appear, it listens first. Then it starts ringing again meaning the
call is not a fax call but a regular call. The pause in the beginning is the concern
where the caller may think there is a problem with the line and may likely end the
call. Or they may get frustrated due to the extra amount of time required for the call
to be setup. This is the biggest concern to keep in mind. You can also use a
dedicated FXO port for FAX calls only to avoid this issue though that is an additional
recurring cost to you. This alternative configuration is shown in step #7.

STEP 6: FAX TO EMAIL DIAL PEER
If the incoming call is a fax call it would be routed to a MMOIP dial peer that is enabled for
our other TCL application that we uploaded. Here it is matching directory number 6700 and
the FAX message received will be converted to a TIFF format and sent to email address
specified in our session target.

dial-peer voice 7 mmoip
description FAX
service fax_on_vfc_onramp_app out-bound
information-type fax
session target mailto:sales@routehub.com

STEP 7: USING A DEDICATED FXO PORT FOR FAX
Here is an alternative solution to the issue we discussed in step #5. Here we would not need
the fax_detect script installed. It just wont be associated to our FXO port for voice calls.
This configuration is similar as seen in the previous steps, but a different application is
applied to this peer. However, most of the configuration is the same.

voice-port 0/1/1
caller-id enable

service onramp
direct-inward-dial
port 0/1/1

description FAX

4.25 Phone Services

We can configure numerous service URLs that users can access, but this is limited and you
want to make sure that Internet access and DNS is all working properly.

These services must be XML based in order for the user to display the content properly on
their IP phone.

Below we will add the common BerBee XML phone service under our telephony-service:

telephony-service
url services http://phone-xml.berbee.com/menu.xml

To access this service once it is added, we would simply press the "Services" button on the
phone to access the XML based service.

Note: Doing this will only allow us to add one phone service to the system. If there
are multiple phone services we want to use then a custom XML service is needed to
list all the other phone services.

4.26 Cisco CME using Exchange 2007 UM

Below reflects the matching configuration on Cisco UC Manager Express (CME) for
connecting with the MS Exchange 2007 UM for voicemail and auto-attendant services. The
connection between the two environments will consist of a SIP trunk (industry standard
protocol). Our CME configuration will exist on the Cisco UC520W appliance.

Our CME configuration will allow SIP to SIP communication and specifying the source
interface for all SIP control and data to be established on our BVI10 interface, the main
interface for our LAN. If you remember in our configuration we added a UM IP Gateway of
10.67.78.1 to our Exchange 2007 UM server, this is the IP address that is configured on our
BVI10 interface (which is shown below for reference).

interface BVI10
ip address 10.67.78.1 255.255.255.0

voice service voip
sip
bind control source-interface BVI10
bind media source-interface BVI10
header-passing

Below reflects the actual SIP trunk configuration to the UM server at 10.67.78.92. You will
see that our destination pattern will be 671. where the dot represents a wildcard mask of
any digit from 0 to 9. This pattern will match our voicemail pilot number (6710) and our Auto
Attendant number (6711). Any call to those numbers will match this dial peer and the route
the call to our UM server.

description EXCH2007-UM
destination-pattern 671.
session transport tcp
dtmf-relay rtp-nte
codec g711alaw

Once that is configured our CME router is now able to use the UM server for voicemail and
AA services. Now, how can we monitor and confirm these operations are working correctly.

The below command is what we can use to validate all SIP based calls through our CME
router. When we execute this command we see there are no calls established.
uc01tra#show sip-ua calls
SIP UAC CALL INFO

Number of SIP User Agent Client(UAC) calls: 0

SIP UAS CALL INFO

Number of SIP User Agent Server(UAS) calls: 0


From 6700, we will call 1001 and the call will go to voicemail, 6710. This will match dial peer
303, which will be routed across the SIP trunk to the UM server. When we execute the same
command we now see more information confirming that our configuration is working. It
shows that our CME router (at 10.67.78.1) has connected with the UM server (at
10.67.78.92) via SIP for an established call. We can also see the Calling (source) and the
Called (destination) numbers in this call. Other details are present reflecting the CODEC and
DTMF used.
SIP UAC CALL INFO

Call 1
SIP Call ID : D78A25CC-86A11DF-898AEA98-
1E376711@sanfrancisco-1.vtnoc.net
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 1001
Called Number : 6710
Bit Flags : 0xC04018 0x100 0x80
CC Call ID : 1172704
Source IP Address (Sig ): 10.67.78.1
Destn SIP Req Addr:Port : [10.67.78.92]:5065
Destn SIP Resp Addr:Port: [10.67.78.92]:5065
Destination Name : 10.67.78.92
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 1172704
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g711alaw (160 bytes)
Codec Payload Type : 8
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
Media Source IP Addr:Port: [10.67.78.1]:16694
Media Dest IP Addr:Port : [10.67.78.92]:60800

Options-Ping ENABLED:NO ACTIVE:NO

SIP UAS CALL INFO


4.27 Using a XML Menu File for Phone Services

On Cisco CME voice engineers can only add one phone service at a given time on the
system. Well if we wanted to add additional services such as BerBee for Weather & News
and the service VoiceView we will be stuck unable to add both.

What do we do then? Simple, we need to create a custom XML file that will include these
two and potential other services, upload this file to a web server then point the phone
services URL to this custom file web location for our phones to use. Here is how we would
do this.

1. First create an XML file, which we will call menu.xml that will consist of two
services: BerBee (for Weather, Stock, and News info) and VoiceView (the ability to
view and access voicemail services directly from the phone display).

menu.xml

<?xml version="1.0" encoding="utf-8" ?>
<CiscoIPPhoneMenu>
<Title>Phone Services</Title>
<Prompt>Please make your selection.</Prompt>
<MenuItem>
<Name>VoiceView</Name>
<URL>http://10.67.5.2/voiceview/common/login.do</URL>
</MenuItem>
<MenuItem>
<Name>Weather,News,Stocks</Name>
<URL>http://phone-xml.berbee.com/menu.xml</URL>
</MenuItem>
</CiscoIPPhoneMenu>

2. We will upload this file (menu.xml) to a web server (in our case we will use
www.routehub.com for our webserver). This file exist today on our server, so
download it if you like.

3. Under telephony-service we will specify our phone service URL for all phones to
use. The name of this service will be called Phone Services, which is what we
should see on our IP Phone. Once we are done dont forget to save the
configuration.

telephony-service
url services http://www.routehub.com/menu.xml Phone Services

4. Last we will restart all IP phones.

telephony-service
restart all


5. Now from one of our IP phones we will go to Services then we will see our created
service called Phone Services listed. When we click on Phone Services we see
our two services listed from the menu.xml file we created.

6. Any new phone services we want to add in the future we can simply just update the
menu.xml file on our web server.

4.28 Installing SIP Firmware on Cisco 7940/7960

When to use: Connecting with a SIP server or provider for Voice IP Communications
using SIP

1. First determine the following about the IP Phone you want to use running the SIP
firmware:

Cisco IP Phone Model (e.g. Cisco 7960)?
MAC Address of Cisco IP Phone?
IP Address of SIP Server (e.g. SIP Provider, Cisco UCM, Cisco CME,
Asterisk)?
Username and Password from SIP Server/Provider?

In our case we will be using the following:

Cisco 7960 IP Phone (CP-7960G)
MAC Address of IP Phone is 0012.00A7.72EA (determined directly on the
bottom of the phone)
Our SIP Server will be our Cisco CME Router (shown in this workbook),
10.67.78.1
Our SIP Username and Password on the Cisco CME Router is: 8778 /
cisco6778.

The extension/number of our phone would be 8778.

2. Next we need to go to the Cisco CCO Download Center and download SIP Firmware
version 6.3 for our Cisco 7960G IP Phone.

This will be a zip file with the name P03-06-3-00.zip.

Once the zip file is downloaded, extract all the content to a folder somewhere on a
computer that will act as out TFTP server. There should be two files:

P0S3-06-3-00.bin
P0S3-06-3-00.sbn

Note: In your case choose the SIP firmware for the correct IP phone that will be
used.

3. Create a new text document called OS79XX.txt containing the SIP firmware image
name:

Note: This file is included in this package for reference.

4. We need to create a new file called "SIPDefault.cnf" that will contain the IP Address
of the SIP server and the image of our SIP firmware among other details:



5. We need to create a new file that is called "SIPmacaddress.cnf" where "mac-
address" would be the MAC address of our IP Phone. Therefore our filename would
be the following:

SIP001200A772EA.cnf

Within this file it will consist of our username, password, and line appearances on our
SIP Server:

As you will see we will use a single line for extension 8778 and we will set our caller-
ID to be "SIP User (8778).


6. Below are the files we should now have all together in a folder:


7. We will launch our TFTP server on our computer using that folder with all our files as
the root folder to source. You can use Solarwinds TFTP application, which is free.

8. We will enable DHCP services on our Cisco UC520 router for VLAN11 (10.67.99.0
/24) that will be used for SIP clients: We will include option 66, which is what SIP
clients will use to find the TFTP server.

ip dhcp pool SNCG-DHCP-VLAN11
network 10.67.99.0 255.255.255.0
option 150 ip 10.67.78.3
dns-server 4.2.2.2
option 66 ip 10.67.78.3

9. Our IP phone will be plugged into port 6 on our Cisco UC520 router assigned to
VLAN11:


10. Below is a summary of the SIP server configuration on our Cisco CME router.

voice service voip

mode cme
max-dn 12
max-pool 12
timezone 47
time-format 24
date-format YY-M-D

voice register dn 2
number 8778
name ROUTEHUB User (SIP)

id mac 0012.00A7.72EA
number 1 dn 2
codec g711ulaw

Note: The step-by-step configuration and explanation is shown within this workbook

11. At this time we will plug our Cisco IP Phone into port 6 (assigned to VLAN11) on our
UC520 router.

The phone will get its IP address via DHCP and look to the TFTP server for any
firmware and it's configuration. Below we can see that activity on our TFTP server.

12. Once our SIP firmware process is completed this is what we should now see on our
phone.

13. From our SIP phone if we call an internal phone this is what they would see:

4.29 VoiceView

VoiceView is a great feature that provide visual voicemail (just like on the iPhone) on your
Cisco IP Phone.

You can view all voicemail messages visually from the Cisco IP phone display including
administer phone greetings among other options.

The configuration for VoiceView is simple and is implemented on both CME and CUE.

1. Requirements:

We will need to know the following info:

The IP Address of CUE: If our case it is 10.67.5.2

2. Enable VoiceView

First we need to access Cisco Unity Express (CUE) and enable VoiceView.

service voiceview
enable
session idletimeout 30
end

3. Authentication URL

Next let's go to our CME router to enable the Authentication URL to allow our users
to administer their voicemail messages and options from the phone. Within the
Authentication URL that is configured under our CME server we will include the IP
Address of CUE as shown:

telephony-service


4. Phone Service

Next we need to configure the phone service for VoiceView that will allow our users
to access their visual voicemail.

We are using a custom XML file for our phone service (entire process shown within
this workbook) that contains our VoiceView phone service. Using a custom XML file
allows us to add multiple phone services.

telephony-service
url services http://www.routehub.com/menu.xml Phone Services

Here is what menu.xml contains which includes the VoiceView service. As you will
see we have included the IP Address of the CUE in the URL.

<?xml version="1.0" encoding="utf-8" ?>
<CiscoIPPhoneMenu>
<Title>Phone Services</Title>
<Prompt>Please make your selection.</Prompt>
<MenuItem>
<Name>VoiceView</Name>
<URL>http://10.67.5.2/voiceview/common/login.do</URL>
</MenuItem>
<MenuItem>
<Name>Weather,News,Stocks</Name>
<URL>http://phone-xml.berbee.com/menu.xml?opts=95377</URL>
</MenuItem>
</CiscoIPPhoneMenu>

OR

We can configure the direct VoiceView phone service under CME if we are not using
a custom XML file. As you will see we have included the IP Address of the CUE in
the URL.

telephony-service
http://10.67.5.2/voiceview/common/login.do


5. Testing

So let's show the process for using VoiceView. From another phone (at extension
1001) we will leave a voicemail for user at 6700.

Here we are using the Cisco IP Communicator at extension 6700. We see a new
message waiting for us.

From our IP phone we will go to "Services"


From there we will select "Phone Services" then "VoiceView"


On our display we need to login using our voicemail (CUE) username and password.
In our case, it would be 6700 and our password is 6700.

From there we are logged in and this is what we see including 1 New Message (seen
under Inbox as 1 N).


To view our voicemail messages we would go to "Inbox" to view all new or saved
messages. Here we see a new message from a caller at 1001. We can go into the
actual message details to listen to the message through our IP Phone.

We can also delete the message from our display or save.

There are other options available for us to use from the main VoiceView display for
administering other voicemail options.

By clicking on Send Message we can send messages to other users on the
voicemail system based on their number:

We can administer our greetings, change our PIN number, to changing our recorded
name under Options.

To sending out broadcast messages to all voicemail users with details of an
emergency or general message.

4.30 Installing SIP Firmware on Cisco 7941/7961/7962+

Installing the SIP firmware for Cisco 7961 and 7962 phones is a little different compared to
the process for installing the SIP firmware for lower-end IP phones like the Cisco 7940 and
7960 series.

But there is no big concern that is the biggest reason why the SIP firmware installation
doesn't work. We will discuss this further within the steps.

Again most of steps are similar to installing the SIP firmware on a Cisco 7940 and 7960 IP
Phone, so you can reference that section at any time.

1. First we need to download the SIP firmware for a Cisco 7961 IP phone from the
Cisco Software Center

We will download the file "cmterm-7941_7961-sip.8-5-4.zip" that is listed on the
page.

2. Next we will extract the files to the desktop of our computer and place into a folder
called TFTPFODLER. These are the files provided in that zip file we downloaded:


3. Next we will start a TFTP server on our computer that will use the folder
"TFTPFOLDER" as the root folder that is located on our desktop containing all the
SIP firmware files.

4. Within the files we downloaded, locate the file "OS79XX.txt". Below are the contents
of that file

We will need to reference "SIP41.8-5-4S" within that file for the next step.

5. This next part is VERY important and the main reason why this process doesn't work
for many people.

We need to create an XML file called "XMLDefault.cnf.xml". This file needs to be in
the TFTPFOLDER folder on our desktop.

This files must be correctly parsed (meaning in the correct format) or the IP Phone
will NOT load the SIP firmware.

This XML file must consist of the following:

Within this file we need to specify the SIP firmware image name that we will use.
The SIP firmware image we will include if what we gathered from step 4, which was
"SIP41.8-5-4S".

So within this sample XML file we will locate the line with the IP Phone we are using,
which is the Cisco 7961 IP Phone, and include the SIP firmware image name within
that line.

Make sure the format is correct and for reference since this is a common issue we
have included the actual XML file with our CME workbook package.

6. At this time we are done. We will plug our 7961 IP Phone into the network, which will
get an IP address via DHCP including knowing that the TFTP server to use for
obtaining it's SIP image is 10.67.78.3, the IP Address on our computer.

Below is what our DHCP server configuration looks like on our Cisco UC520 router:

ip dhcp pool SNCG-DHCP-VLAN11
network 10.67.99.0 255.255.255.0
option 150 ip 10.67.78.3
dns-server 4.2.2.2
option 66 ip 10.67.78.3

7. The IP phone will find the TFTP server will start installing the SIP firmware. We will
see this download and install process directly from the phone display.

Once it is done we can go into the phone to make all the necessary SIP configuration
(server, account) under Settings.

4.31 Cisco Unified CallConnector

Cisco Unified CallConnector (UCC) is a solution that allows users to use their phone features
like placing calls from their computer via their Outlook client or web browser. The call will be
placed through the users actual physical phone.

UCC is only supported on Cisco CallManager Express (CME) routers.

UCC is installed as a server and as a client. UCC server connects to a Cisco CallManager
Express (CME) router where it pulls the entire configuration via Telnet. It uses that
configuration to understand what directory numbers and IP Phones (ephone) exist on the
CME router. User accounts are created on the UCC server then associated an existing IP
Phone (ephone) and directory number on the CME router. The user account can then be
used by the user to login and use the UCC capabilities.

On the UCC client end, a program is installed that places a tool bar menu in Internet Explorer
and Outlook (if installed). From this tool bar we can place calls, setup a conference call,
forward calls, and use other phones features on CME.

The downside to this solution is that it adds an additional cost with licensing and software to
use UCC. Meaning it is not free to use. Plus it is another device (server) on your network
that needs to be managed and maintained.

The software can be found on the Cisco Software Center and the IP address of our UCC
server (installed on Windows 2003 server) will be 10.67.78.194 that will connect to our CME
router (10.67.78.1). That diagram is found in this package and reference for most of the
features shown in this workbook.

4.31.1 Server Installation

Cisco Unified CallConnector (UCC) will be installed on a Windows 2003 server (in a virtual
machine). Installing on a virtual machine is not supported, but we will show you what is
needed to successfully install and configure UCC in our voice environment.

1. Once we download the installation file for Cisco Unified CallConnector lets go ahead
and start the installer.

2. FAIL! We are not allowed to install Cisco Unified CallConnector a virtual machine
and the installation will stop

3. We will go ahead and stop our Windows 2003 server virtual machine and we will
make the following changes to our VMware file for this virtual machine to bypass this
warning and continue with our installation.

Note: This step should only be used for development and testing purposes. For a
production environment Cisco Unified CallConnector should be installed on a Cisco
approved physical server.


4. Once we restart our virtual machine then restart the installation wizard again we are
successful to continue with the install process. We will continue through the
installation wizard:



During the installation it will ask about the Licensing Option. We can input the PAK-
ID or serial number if we purchased the necessary licensing for our production
environment. Or we can try Cisco Unified CallConnector for 45 days. We will use
the 45 day trial for our deployment for testing purposes especially since its installed
on a non-Cisco approved system.

Selecting the 45 day trial will prompt us with the following warning.

5. Once the installation is completed it will start the Step Wizard for us to complete (or
confirm) the following tasks. Its best to follow the steps provided starting by clicking
on Next which will go through each option. We also want to make sure Show only
Basic Configuration Pages is checked. If we uncheck that box then the gray-out
options such as System Tracker or Database Server can be available for
configuration. Going through the basic setup is more than sufficient to get Cisco
Unified CallConnector functional in our Cisco CME environment.

Therefore, click Next to continue.


6. On the next window it will confirm what licensing is installed and will be used. We
also have the option of setting up a new license and activating it from this page.

7. Next it will ask us to specify and test our SIP server parameters. We want to confirm
that our Windows 2003 server SIP domain is listed (IP address of our server)
including the SIP port number (should be 5060). We want to confirm that the SIP
port is listening and working on our server by clicking on Test Port. We will leave
everything else listed at their defaults.

8. On the next page we want to do a few things here, but essentially we want to add the
IP address of our Cisco CME router including the username and password needed
for logging into the Cisco router via Telnet. We also want to do the following on this
page:
We will include our voicemail number 6000 under Transfer to voicemail
mail number
Our extension schema uses 4 digits for the extension.
To place any external call we need to place a 9 at the start of all calls
made.
Before we can continue with the next page we need to specify the
CME/UC500 location. We can do that by clicking on Edit.

9. A new window will open where we will input the name to be USA and include:

Country: United States
Area Code: xxx
Number Pattern: (xxx) xxx-xxxx

Once we are done we want to make sure we that we Add that to the Location
Name table then click OK.

10. On the main page we now see our CME/UC500 location details listed. Once we are
done we need to click on Add to add our Cisco UC520 device to the CME/UC500
list. Then we can click Next to continue.

11. On the next page it will connect to our CME router (10.67.78.2) via Telnet to pull the
configuration that consist of all the extensions and dial plans necessary for the next
steps and using the CallConnector service. We would click on Start that will
connect and start pulling the configuration. Once it is done we can click Next.

12. During this process it will display a warning that some of the router configuration
requires the EXEC (privilege mode) password. We will click Yes to continue. Make
sure that password Exec Password field is filled in (if applicable). Our user account
mthomati has the privilege password already configured on the router.

13. We see that it is downloading the data and updating its database.

14. On the next page within the left table we see all the Ephones listed for our added
CME router. We can also see which devices are currently registered. We will setup
a new user account for CallConnector. We will select the device that is using
directory numbers 6700 and 7700. That is listed as ID 6. When we select that ID
we will click on Next.

15. The next page deals with groups that our users can be added to including what
access to other groups users can see. For example, we have added a new group
called Sales. Users added to the Sales group can view other users that may be
added to groups like Marketing or Finance. We have added two groups here, Sales
and Support.

16. On the next page we will add a new user that will use the username routehub that
will belong to the group Support. For the user type, we will specify User. Make
sure that the account is updated and added (shown on the left table) before we
continue.


17. On the next page we will associate our selected ephone to our new user account.

18. On the next page we can setup email notifications for announcements or alerts.
Here we will setup all of our SMTP server parameters.

Note: In order for users to receive emails, the email address should be listed for our
user account during setup.

19. We can send a test email to confirm that our Cisco CallConnector server can send
out emails correctly.

20. On the next page this is where we can send problem reports that can include log
attachments for SIP or logs for the database server as an example.

21. On the next page we are finished with the setup wizard that mainly consist of adding
our CME router, adding user accounts, and groups. We can always re-run this setup
wizard to add additional users and groups when needed. When we are ready to
commit to the changes we will click on Finish.

22. A window will appear stating that it will save the configuration and that it will take a
couple of minutes.

23. For any new changes that we make we need to restart our services. We will click OK
and not worry about this step for now because we will need to reboot the server
soon.

24. At this time the installation and the setup for CallConnector is completed. We will
proceed and reboot our computer.

4.31.2 Components

Once UCC is installed and configured its important to understand what is installed on the
UCC server including what to check to confirm that UCC is working.

1. Once Cisco Unified CallConnector is installed on our Windows 2003 server it can be
found under our Program list:


2. When we go to Cisco Unified CallConnector Service Manager listed under the
program folder we need to confirm that all services listed here are Started before
we start testing the operation for CallConnector.

4.31.3 Adding a new user

Once UCC is configured, on-going configuration will mainly include adding new user
accounts and then associating their IP phone with their account.

1. If we want to add a new user account to our CallConnector server we need to rerun
the Startup Wizard which can be found under programs. From this page we will
click Next to continue. Nothing needs to be changed here.

2. Under the Select Ephones page we will select and add the ephone with directory
numbers 6700 & 1002 (found as ID 2).

3. On the next page we can verify the ephone that we recently added to our
CallConnector server.

4. On the next page we can view the actual directory numbers and call appearance for
the ephone.

5. On the next page we will add a new user account called ruser that will be part of the
Customer support group. We also see the other user account listed under the
User Table:

6. And last we will associate the ephone (and directory) details that will apply for this
account for controlling their IP phone from the CallConnector client software.

We would repeat this process for additional users we want to add.

4.31.4 Client Installation

1. Next we will install the Cisco Unified CallConnector software on our Windows XP
client machine.

2. Next the following window will appear telling us to close all instances of other
applications running on our machine before the installation wizard can continue. We
will do that now then continue.


3. Now the installation will continue.


4. On the next page we will specify our customer information

5. On the next page we will specify the IP address for our Cisco CallConnector server
(10.67.78.194) including our user account routehub that was added to our server
associated with our IP phone using directory number 6700 & 7700. We also want to
use CallConnector with our MS Outlook program, so on our XP computer where
Outlook is installed we will specify the Email profile and the password we use to login
to our email.


6. From here we will continue with the installation process.

7. On the next page, just like our server, we need to either input our license or we can
use a 45-day trial to use the software. We will use the 45 day demo for our
environment.


8. Once the installation is completed we need to restart our computer.

4.31.5 Using Cisco Unified CallConnector

1. When the CallConnector client program is installed on our computer we should see a
desktop shortcut called CallConnector Popup. We will click on the shortcut to run
the program on our computer.

2. We should see CallConnector running on our computer now, which is seen on the
menu bar. When we right-click on that shortcut and we can go into properties for
additional details for configuration.


3. Under the Server tab this is where we see the UCC server IP address including
user account for logging into the server. From this tab we can see whether we are
Connected to the UCC server and Database server. This is a great place to go first
on our client machine to confirm that we are connected to the UCC successfully.
Any changes that we do here we need to make sure to Apply the changes.

4. Under the Dial Plan tab this is where we would add our location and patterns for
placing calls through CallConnector. We will include our area code, 209, which is
where our CME router is located with a 209 DID.

5. Lets quickly discuss some other aspects for UCC. From the preferences page within
the UCC pop-up we can confirm if our client is connected to the UCC server. We
can also do this from our Outlook client under Connection Status.

6. Now lets show how we can use UCC starting with our MS Outlook 2007 client.
There we see a tool bar listed. We want to ensure that we are connected, which is
seen as the green circle on the tool bar. From this tool bar we can place calls,
pickup calls, or even forward calls from this menu.

7. When we go to Contacts then Contacts again we can see all users (added to
UCC) listed including if they are online or offline. Since we added two accounts, we
only see two including our account which shows online


8. From the Contacts page when we go to Status we can view users that are online or
offline (like under Contacts) but we can see them based on the groups they are
assigned to. Remember we have our user account routehub in the group Support
and our other account ruser in group Customer support.

9. We can also use UCC within our web browser (Internet Explorer), which has the
same tool bar.

10. One of the great uses with using UCC with the browser is that any number on a web
page we can right click on that number then dial that number directly to our IP
phone (associated with our user account). Or there are other options like creating a
speed dial or Outlook contact.

11. Now lets do a test to show how this works. From our tool bar we want to dial
extension 1002 to a user within our internal network.


12. From the UCC pop-up we can see that our phone is placing a call to 1002.

Note: When the call is answered the call is handled from the actual physical server
not the computer. Meaning we should not be talking into the computer to
communicate with the user at 1002. This call is handled from the phone.


13. Below are some of the call features we can do from our UCC client (via Outlook or
the web browser).

5 Configuration for CUE

5.1 Access to CUE

Configuring Cisco Unity Express (CUE) for the first time requires us to console into CUE from
our CME router. CUE may exist in different form factors, but its an extra component that is
required to be added to the CME router. They can exist as the following:

AIM card
Network Module (NM)

An Advanced Integrated Module (AIM) is a like a small daughter card with a Cisco Flash that
is installed into one of the AIM ports inside the router. This can be used for Cisco ISR 2801
series routers up to the Cisco ISR 3800 series including the Cisco 3700 series router, but the
voicemail storage space is small.

CUE can also exist as a Network Module (NM) to provide higher number of voice mailboxes,
mailbox sizes, and overall storage ideal for larger SMB environments. This requires Cisco
routers that support network modules like the Cisco ISR 2811 series and higher.

Regardless what is installed with the CME router access to the CUE console is the same,
you just need to know what the service module name and port number is before continuing.

5.1.1 CME Configuration

First we need to configure the Service Module on our CME router in order to console into the
CUE.

STEP 1: LOCATE THE SERVICE MODULE NAME
There are many ways to do this and will likely be easy enough where you can skip this step.
But I like to do a show ip interface brief to display all interfaces.

uc01tra#show ip interface brief
Interface IP-Address OK? Method Status Protocol
FastEthernet0/0 1.1.1.73 YES NVRAM up up
In0/0 unassigned YES TFTP up up
FastEthernet0/1/0 unassigned YES unset up up

There I see that my service module is In0/0 (for Integrated Service Engine).

STEP 2: SERVICE ENGINE IP CONFIGURATION
Next we will go into our config mode and extend (or bridge) the subnet used on our LAN
interface to our service engine to share. In our configuration we will use the IP address that
is configured for BVI 10:

interface Integrated-Service-Engine0/0
description ROUTEHUB: CUE interface
ip unnumbered BVI10

As a recap this is what our BVI10 interface is configured for:

interface BVI10
ip address 10.67.78.1 255.255.255.0

Continuing under our Integrated-Service-Engine interface we will enable NAT to allow our
CUE to access external resources like sending emails when that feature is configured. We
will specify the IP address for the CUE service engine including what default gateway IP it
would use, which would be the same IP as our IP unnumbered. Below is that continued
configuration:

ip nat inside
ip virtual-reassembly
service-module ip address 10.67.78.2 255.255.255.0
service-module ip default-gateway 10.67.78.1

STEP 3: STATIC ROUTE TO CUE
Last we need to configure a static route for the IP address we configured for the CUE module
where the next-hop is the service engine itself. Without this configuration we will be unable
to route or management to this CUE service engine.

ip route 10.67.78.2 255.255.255.255 Integrated-Service-Engine0/0

5.1.2 Console into the CUE Service Engine.

To access the command prompt of the CUE service engine, we need to do a reverse telnet
into CUE service module based on the interface name we determined from a previous step.

uc01tra#service-module integrated-Service-Engine 0/0 session

-----------------------------------------------------------------------
Powered by...
|| ||
|| ||
|||| ||||
..:||||||:..:||||||:..
c i s c o S y s t e m s
OFFICAL USE ONLY!
RouteHub Group, LLC
(925) 230-2203
www.routehub.com
support@routehub.com
-----------------------------------------------------------------------

User Access Verification

Username: routehub
Password:
cue01tra#
cue01tra#

NOTE: For first time CUE configuration there is no username/password or enable
passwords configured, so you can simply type in enable at the CUE user prompt to
get into the enable mode to start the configuration.

5.2 Unity Express

Cisco Unity Express (CUE) is a voicemail application solution aimed for Small/SMB
businesses. They provide many voice features that include voicemail, auto attendant,
sending voicemail messages via email, Live Record, and more. Voicemail is the most
common and critical component in the Unified Communication network infrastructure by
providing the same functions as email messaging provides within data networks.

Before any of the features within this CUE Configuration section can be configured we need
to enable CUE initially.

1. First we need to setup CUE globally
2. Enable Voicemail Services including MWI
3. Create User Voice Mailboxes
4. Enable Other CUE Services like Auto Attendant

5.2.1 CUE Global Configuration

Below is the basic configuration we need to enable on our CUE service engine globally
before we enable some of our CUE services like Voicemail.

STEP 1: GENERAL CONFIGURATION
First lets configure what our hostname will be including our domain name, time-zone, and
preferred system language to use for all CUE greetings and messages for the system:

hostname cue01tra

ip domain-name routehub.com

clock timezone America/Los_Angeles

system language preferred "en_US"

STEP 2: ADMIN USERNAME AND GROUPS
Likely our admin accounts are automatically configured, if not below shows how we can
create a new user called admin and applying our user admin into a default group called
Administrators.

username admin create
groupname Administrators member admin

We can create other groups if needed for our users to belong in called Users:

groupname Users create

STEP 3: SIP TRUNK CONFIGURATION TO CME
Next configure (or confirm) that the SIP connection to CME is configured. For this
configuration we want to specify the default gateway IP CUE would use for routing calls,
voicemails, and email notifications outside of the CUE service engine. In our configuration,
we will configure the gateway IP to be 10.67.78.1, something we also configured under our
Integrated Service Engine in CME.

ccn subsystem sip
gateway address "10.67.78.1"
end subsystem

5.2.2 Enable Voicemail Services

To enable voicemail services we need to do two things, first we need to enable voicemail and
MWI services on CUE then apply a directory number (the voicemail pilot number) to the
application.

STEP 1: VOICEMAIL APPLICATION
First we need to define the voicemail application, which will likely be already configured on
CUE. You can do so by doing a show run to confirm that the following configuration is
similar.

ccn application voicemail
description "voicemail"
enabled
maxsessions 6
script "voicebrowser.aef"
parameter "logoutUri"
"http://localhost/voicemail/vxmlscripts/mbxLogout.jsp"
parameter "uri" "http://localhost/voicemail/vxmlscripts/login.vxml"
end application

This configuration specifies the script (aef format) that will run the voicemail services when it
is in use). By default it will allow up to 6 sessions to the CUE service engine for voicemail
services, meaning up to 6 people can leave a voicemail messages or messages checked by
users on the network.

STEP 2: CONFIGURE CALL HANDLING (TRIGGER) FOR VOICEMAIL APPLICATION
Next we will specify the phone number, our pilot number that will be used for forwarding calls
to voicemail and accessing the voicemail system for checking new messages. In our
configuration, our voicemail pilot number will be 6000.

ccn trigger sip phonenumber 6000
application "voicemail"
enabled
maxsessions 6
end trigger

STEP 3: MWI APPLICATION
MWI is configured to provide notification lights on a users phone when a new voicemail
message has arrived. First, like what we did with the Voicemail application we will enable our
MWI application. In this configuration we will specify the MWI DN for when a notification is
turned ON (8000) or when the notification is turned OFF (8001).

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

5.2.3 Sending Calls to Voicemail on CME

This configuration on CME is needed for sending calls to a voicemail system via a SIP
connection particularly to a CUE service engine. But, we want to show you the necessary
configuration needed for calls to be sent to voicemail (to Cisco Unity Express) if there is no
answer to a particular directory number.

STEP 1: DIAL PEER CONFIGURATION
First we need to configure a dial peer to match our voicemail pilot number (this is the number
where we want to send calls to voicemail and/or to access our voicemails). In our
configuration that would be DN 6000. This directory number would be our destination pattern
that would be forwarded to the CUE service engine found at IP address 192.168.5.2, which
will be a SIP trunk connection.

codec g711ulaw
no vad

STEP 2: VOICEMAIL BUTTON CONFIGURATION ON CME
Next we will add our voicemail pilot number of 6000 under our CME telephony service. This
configuration will setup a direct speed dial to access our voicemail. When we press the
Mail button on our phone it will dial the voicemail pilot number.

telephony-service
voicemail 6000

STEP 3: SENDING CALLS TO VOICEMAIL ON CME
When the line (or directory number) is busy or is not answered the call would be forwarded to
voicemail. In our configuration if someone is calling 6700, but the line is busy or there is no
answer (within 15 seconds) then the call would be forwarded to voicemail at directory number
6000, which will match the dial peer we configured in step 1.


STEP 4: CONFIGURE MWI
Next we need to configure our Message Waiting Indictor (MWI). This means that if a new
voicemail message has arrived, CUE will send a MWI ON message to the number where the
message was left. A red light would turn on the phone. Once the voicemail message is read
and no longer new then a MWI OFF message would be sent to turn off the red light.

In our configuration, our MWI ON directory number will be 8000 and our MWI OFF directory
number will be 8001. You will also notice that the MWI directory number include . (four

dots) which represents the directory number that is receiving the MWI message. So, if a
voicemail message is left for 6700 then the following MWI message is sent: 80006700. Once
the voicemail message is heard and no longer new then the following MWI is sent:
80016700.

ephone-dn 20
mwi on

ephone-dn 21
mwi off

5.2.4 Create User Voice Mailboxes

Now it is time to create the user voice mailboxes that has IP phones registered with the local
CME router.

STEP 1: CREATE USER ACCOUNT
First we need to create the user account that is a name (following a type of standard), but this
is NOT defining what the DN is for this user yet. This is only creating the user account. In
our configuration, we will create a user account called dn6700 that will have Directory
Number 6700.

username dn6700 create

STEP 2: ASSIGN USER ACCOUNT TO A GROUP
If different groups are configured during the global configuration then that account can be
added to that group or they can be added to the administrator group. In our configuration we
will add our created user under the Administrators group.

groupname Administrators member dn6700

NOTE: As a best practice it is best to place all users in a Users group.

STEP 3: ASSOCIATED DIRECTORY NUMBER (DN) WITH USER ACCOUNT
Now we will associate our DN to our created user account. In our configuration the DN for
user dn6700 is 6700.

username dn6700 phonenumber "6700"

Here is another example where we created user routehub which is associated with the
internal DN of 7700 plus the full E164 number (but this is not required in most deployments):

username routehub phonenumber "7700"
username routehub phonenumberE164 "19252302203"


STEP 4: CREATE USER VOICE MAILBOX
Last we will configure our user voice mailbox. Our user mailbox can support up to 420
seconds for all total voicemail messages. The maximum size for a single voicemail message
can be 60 seconds.

voicemail mailbox owner "dn6700" size 420
description "User DN6700 mailbox"
messagesize 60
end mailbox

STEP 5: DEFAULT VOICEMAIL SETTINGS
Next we can also define what our default settings would be for new mailboxes added to CUE.

voicemail callerid
voicemail default language en_US
voicemail default mailboxsize 420
voicemail broadcast recording time 300
voicemail default messagesize 240
voicemail notification restriction msg-notification
voicemail operator telephone 0

STEP 6: CONFIGURE FULL NAME TO SUPPORT DIAL-BY-NAME
At this time exit the configuration mode by typing in end. Next for our recently configured
mailbox for user at extension 6700 we will define the full number for this user and number.
This is important especially if our Auto Attendant allows callers to dial-by-name.

In our configuration, the full name for the user at extension 6700 will be RouteHub Group.

username dn6700 fullname first Routehub last Group display "RouteHub Group"
password cisco6778

5.2.5 Enable other CUE services (like Auto Attendant)

Follow the configuration for Auto Attendant in section 5.5

5.3 Upgrade CUE to Version 7.x

The process for upgrading the software on the CUE is a more different process compared to
upgrading the software for CME. We will show the steps necessary for successfully
upgrading the software on the CUE service engine.

It is recommended to upgrade CUE to the latest OS to fully take advantage of creating on-
the-fly aef scripts which is extremely easy compared to older CUE versions like 2.3 where it
requires us to use a CUE script editing tool and knowing how to create custom scripts. With
Cisco Unity Express Version 7.0, the web interface allows up to create easy AA scripts
without requiring any AEF scripting knowledge.

Also upgrading to Cisco Unity Express Version 7.0 allows additional new features and
capabilities over version 2.3.

We will show you the steps for upgrading the CUE OS.

NOTE: Before doing an upgrade to a new version it is best to backup your
configuration files and read through all voicemail messages saved on the system
unless this is a new deployment.

STEP 1: CONFIGURE DEFAULT FTP SERVER LOCATION
CUE uses FTP to upload files for the upload process because there are a lot of files involved.
Therefore we need to configure the FTP path and account info on CUE before any upgrade
can occur (though this is not necessarily required). In our configuration we will specify the IP
address of the FTP server including the directory name (if any) that is called cue7. Next we
will specify the username and password on the FTP server that CUE will use for
authentication.

software download server url ftp://10.67.78.243/cue7 username admin password cisco123

STEP 2: DOWNLOAD NECESSARY CUE FILES
Next, to upgrade to Cisco Unity Express Version 7.0 from CUE version 2.3, we need to
download three set of files from the Cisco.com software center.

For reference the CUE files can be downloaded here:
http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml

Since we are installing CUE7 on our UC520 we will download the following files.

The CUE zip file: cue-cm-k9.ise.7.0.1.zip
The Language pack: cue-vm-en_US-langpack.ise.7.0.1.prt1
The License file: cue-vm-license_50mbx_cme_7.0.1.pkg

NOTE: refer to Steps 5A and 5B for information relating to how to determine the correct
license file and how to download the correct CUE files based on CUE hardware that is
installed.

We would copy these files to our FTP server and folder location (under folder cue7) that we
configured in Step 1.

We would unzip the file cue-cm-k9.ise.7.0.1.zip in the same folder with our language pack
and licensing files. One of the files within our zip will be file: cue-vm-k9.ise.7.0.1.pkg

These files should be in our cue7 folder on the FTP server (10.67.78.243):

STEP 3: DOWNLOAD FILES TO CUE
Next from our CUE CLI prompt we will download our CUE files from our FTP server to the
CUE service engine.

software download upgrade cue-vm-k9.ise.7.0.1.pkg

Issuing that command will start displaying the following results:

WARNING:: This command will download the necessary software to
WARNING:: complete an upgrade. It is recommended that a backup be done
WARNING:: before installing software.
Would you like to continue? [n] y
Downloading software install upgrade cue-vm-k9.ise.7.0.1.pkg
Bytes downloaded : 62528
Validating package signature ... done
Validating installed manifests ..........complete.

To confirm that the software download is successful we can issue the following command:

cue01tra# software download status
Download request completed successfully.


STEP 4: INSTALL CUE SOFTWARE
Now we can install the CUE package software we downloaded from our FTP server.

software install upgrade cue-vm-k9.ise.7.0.1.pkg

Issuing that command will start displaying the following results:

WARNING:: This command will install the necessary software to
WARNING:: complete an upgrade. It is recommended that a backup be done
Would you like to continue? [n] y

STEP 5: CONFIRM CUE INSTALL AND NEW VERSION
At this time everything should be successfully installed and we can confirm by issuing the
following command:

cue01tra# show software versions
Cisco Unity Express version (7.0.1)
Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008
by Cisco Systems, Inc.

Components:

- CUE Voicemail Language Support version 7.0.1.0

cue01tra#

STEP 5A: DETERMINE LICENSING
How do we know the correct license file? In our example we downloaded license file cue-vm-
license_50mbx_cme_7.0.1.pkg that reflects 50 mailboxes. Why 50 and how can I confirm 50
on any CUE module.

Simple. On the CUE CLI execute the following command:

cue01tra# show software licenses
Core:
- Application mode: CCME
- Total usable system ports: 6

Voicemail/Auto Attendant:
- Max system mailbox capacity time: 840
- Default # of general delivery mailboxes: 15
- Default # of personal mailboxes: 50

- Max # of configurable mailboxes: 65

Languages:
- Max installed languages: 1
- Max enabled languages: 1

There we see Default # of personal mailboxes: 50 that reflects our 50 mailboxes.


STEP 5B: INSTALLING CUE SOFTWARE FOR OTHER CUE MODULE TYPES
This process is similar for CUE on AIM, NME, or NM. The only difference is the file CUE type
(e.g. AIM, NM) will be listed in the name than ISE which is for the UC520.

For example, we have a Cisco UC520 appliance. One of the files we downloaded was:

cue-cm-k9.ise.7.0.1.zip

Why that file and how do we know that is for the UC520? There are two parts in this file and
the other two core files we downloaded; the platform and the version.

Where you see ise is the platform and where you see 7.0.1 is the version.

ISE pertains to UC520 hardware.

Other platform types may include the following:

NME: for NME-CUE
NM-AIM: for NM-CUE, NM-CUE-EC, and AIM-CUE
ISE for UC520

So if our CUE resides on AIM-CUE and we wanted version 7.0.2 (as an example) then one of
the filenames would be:

cue-cm-k9.nm-aim.7.0.2.zip

5.4 Coping Files to CUE via CLI

To copy files to the CUE service engine is different compared to how we usually upload files
with Cisco routers or firewalls.

It requires having an FTP or HTTP server where the source file (script file or prompt file)
exist.

Here we are copying an AA prompt audio file from our FTP server to our CUE service engine.

ccn copy url ftp://10.100.10.123/AAprompt1.wav prompt AAprompt1.wav

5.5 Auto Attendant

Auto Attendant (AA) is a feature used within Cisco Unity Express (CUE) that provides an
automated menu system for callers to obtain basic information like the address and business
hours for the company including calling users based on their extension. This is common if
everyone doesnt have a dedicated phone number especially for many small networks.

STEP 1: CME CONFIGURATION
There are many ways to do this, but the best way is to use AA with multiple analog lines
plugged into FXO ports setup for PLAR. PLAR is a feature that will automatically dial an
extension once that FXO port is receiving a new call. In our configuration, the PLAR would
be to the AA number, which will be 6003.

voice-port 0/0/3
pre-dial-delay 0
no vad
connection plar 6003
caller-id enable

After the voice-port is configured, all calls to 6003 need to be routed to the CUE service
engine, which is a SIP connection from our CME router to CUE. The IP address for our CUE
service engine would be 10.67.78.2. We will match any number from 6000 to 6999 as this
directory schema will be dedicated for CUE services like voicemail and AA. Hence, our AA
DN of 6003 would fall within this range.

destination-pattern 6...
codec g711ulaw
no vad

STEP 2: CONSOLE INTO THE CUE
After our CME configuration is completed from our CME router we need to console (or telnet)
into our CUE module. Since our CME router is a Cisco UC520, our CUE module is located
on the In0/0. So we will session into that service module:

uc01tra#service-module integrated-Service-Engine 0/0 session

-----------------------------------------------------------------------
Powered by...
|| ||
|| ||
|||| ||||
..:||||||:..:||||||:..
OFFICAL USE ONLY!
RouteHub Group, LLC
(925) 230-2203
www.routehub.com

-----------------------------------------------------------------------

User Access Verification

Username: routehub
Password:
cue01tra#
cue01tra#

STEP 3: AA APPLICATION (CUE)
Next we need to enable the AA application on CUE. The AA, by default uses a default script
called aa.aef. Advanced CUE AA scripts can be setup with certain parameters, uploaded to
the CUE, and applied to the AA application. This will be a separate AA section we will
provide soon. This is a default script that should already be configured. In this AA
configuration, a default greeting will be present where the caller can dial by extension or by
name. Below is that default AA configuration under our CUE service engine.

ccn application autoattendant aa
description "autoattendant"
enabled
maxsessions 6
script "aa.aef"
parameter "busClosedPrompt" "AABusinessClosed.wav"
parameter "holidayPrompt" "AAHolidayPrompt.wav"
parameter "welcomePrompt" "AAWelcome.wav"
parameter "disconnectAfterMenu" "false"
parameter "dialByFirstName" "false"
parameter "allowExternalTransfers" "false"
parameter "MaxRetry" "3"
parameter "dialByExtnAnytime" "false"
parameter "busOpenPrompt" "AABusinessOpen.wav"
parameter "businessSchedule" "systemschedule"
parameter "dialByExtnAnytimeInputLength" "4"
parameter "operExtn" "0"
end application

STEP 4: AA CALL HANDLE/TRIGGER (CUE)
Once the AA application has been enabled and configured (or confirmed) we need to setup
our trigger. The trigger is where we specify the directory number that would be associated to
our AA including the number of AA sessions (or calls to AA) that are supported. In our
configuration, our AA number will be 6001 and we will support up to 5 sessions.

application "autoattendant"
enabled
locale "en_US"
maxsessions 4
end trigger

STEP 5: USING AA
Use a caller calls the full phone number (DID number) that is associated with AA DN of 6003.
The call would be routed to the CUE based on VOIP dial peer 600. There they will hear a
general greeting where caller can dial a person by either there name or extension. They also
have the option of pressing 0 to access the operator.

STEP 6: ENABLE PROMPT APPLICATION AND TRIGGER
You are likely asking right about now, how can you create a custom prompt than using the
default. Its pretty easy, we need to enable the Prompt Manager application then configure a
trigger using an unused directory number like 6006 in our configuration:

ccn application promptmgmt
description "promptmgmt"
enabled
maxsessions 1
script "promptmgmt.aef"
end application

application "promptmgmt"
enabled
idletimeout 5000
locale "en_US"
maxsessions 1
end trigger

STEP 7: USING THE PROMPT MANAGER
Now from one of our IP Phones we can simply dial the prompt manager DN, 6006, that will
provide a menu for creating custom greetings.

For example, lets say we dial 6006. It will ask for us to enter our extension so lets input one
of our extensions, 6700 followed by our PIN number (configured once we setup our voicemail
greeting and PIN number from our phone).

Next it will ask what we want to do, we will press 1 to administer the AA greeting.

From there we will be able to record our new greeting then active it. Thats it! Now when
callers call into the AA they will hear the custom greeting.

One thing to keep in mind, the greeting must be align with what options are available. For
example, in the default AA, we can only press 1 to dial by extension, press 2 to dial by
name, or press 0 to contact the operator. Your custom greeting must include those options.
If you want different actions for what to do when pressing a button then again a custom AA
script is required and will be included in this CDCM workbook soon.

5.6 Voicemail Email Notifications

Make sure to have the latest CUE software running before implementing email notifications
when a new voicemail arrives.

This is close to what we call Unified Communications in some sense, but when a person
leaves a voicemail message your phones notification light will turn red. Well a common
question is can that voicemail message that is left on the phone be sent as an attachment via
email. So they can listen to the voicemail message from our iPhone, Blackberry, or
computer.

In this configuration, any voicemail message left for number 6700 will also be sent via email
to vm@routehub.com.

STEP 1: ENABLE VOICEMAIL NOTIFICATION
By default voicemail notification is disabled, so we need to enable voicemail notification on
CUE including support for email attachments, which would be our voicemail message that
would be a wav file.

voicemail notification enable
voicemail notification preference all
voicemail notification allow-login
voicemail notification email attach

STEP 2: SPECIFY SMTP SERVER AND ANY AUTHENTICATION REQUIRED
Next we need to specify the hostname or IP address of our SMTP server that CUE will
communicate with for sending voicemail messages via email. Some SMTP servers
especially internally do not require any authentication for sending emails. In our
environment, no authentication is needed. The IP address of our SMTP server is 10.67.78.6.

smtp server address 10.67.78.6 authentication none

STEP 3: SPECIFY THE FROM EMAIL ADDRESS
Next we will specify what the From address would be for our email messages. This is the
email address that we would see as From for the voicemail emails we will receive. In our
configuration we will say the From email address is support@routehub.com.

voicemail configuration outgoing-email from-address support@routehub.com

STEP 4: ENABLE VOICEMAIL NOTIFICATION FOR USER WITH NUMBER 6700
Next we need to enable voicemail notification for the user or users where their voicemail
messages will be sent to them. In our configuration we will use the user at DN 6700 which is
dn6700.

As a recap, that user was created and associated to directory number 6700 as followed:

username dn6700 phonenumber 6700

Below is the configuration needed for enabling dn6700 for voicemail notifications.

NOTE: This configuration is not done at the config mode, but at the enable mode.
Therefore, if you are at the config mode prompt type in end then continue below.

voicemail notification owner dn6700 enable

STEP 5: SETUP VOICEMAIL PROFILE AND SCHEDULE FOR USER
Now we will setup our voicemail profile (called VM-6700) for our user dn6700. Any voicemail
left will be on our phone, but we will also receive an email (sent to vm@routehub.com) where
we can listen to our voicemail message anywhere than being restricted to our physical
phone. The voicemail message will be attached. Our profile will have an active schedule of
24 hours 7 days a week. Meaning we will always get an email even 3AM if a new voicemail
message is left:

NOTE: This configuration is not done at the config mode, but at the enable mode.
Therefore, if you are at the config mode prompt type in end then continue below.

username dn6700 profile VM-6700 email address vm@routehub.com
username dn6700 profile VM-6700 email enable
username dn6700 profile VM-6700 email preference all
username dn6700 profile VM-6700 email attach
username dn6700 profile VM-6700 email schedule day 1 active from 01:00 to 24:00

VERIFICATION
We can confirm and view our voicemail notification configuration for a user with these two
commands (where X is the username ID like in our previous example it would be dn6700)
show voicemail notification owner X profile
show voicemail notification owner X email

5.7 CUE and CME on separate routers

In this configuration we will show the necessary configuration needed if CME is on one router
and CUE is on another router.

First, lets explain the environment. Our IP Phones are registered with the CME router using
1XX directory numbers. We will configure only one phone with DN 199, which will be our test
user. On our CUE router, our voicemail (VM) pilot number will be 2000, AA will be 2001, and
our Prompt manager (PM) will be 2002. Our MWI directory numbers for ON will be 800XXX
and OFF will be 801XXX where XXX is our three digit user extension.

5.7.1 CME Router

STEP 1: ALLOW VOICE PROTOCOL CONNECTIONS
First lets specify all SIP and H.323 communications are allowed to/from our CME router.

voice service voip

STEP 2: VOIP DIAL PEER FOR CUE SERVICES (VM, AA, PM)
Next lets configure a SIP trunk matching the 2XXX directory numbers (any DN from 2000 to
2999) pointing to the CUE service engine (at 192.168.1.2).

codec g711ulaw
no vad

STEP 3: VOIP DIAL PEER AND DN FOR MWI (ON, OFF)
Next we want to configure a dial peer that will match any incoming call that matches the MWI
numbers from the remote CUE service engine.

incoming called-number 80[0,1]...
codec g711ulaw

We will also specify the MWI ON and OFF DN on our CME router to create MWI lights on our
registered phones when a new message arrived or is read.

ephone-dn 20
number 800...
mwi on

ephone-dn 21
number 801...
mwi off

STEP 4: PHONE AND DN CONFIGURATION
Next we will configure the DN of 199 that will be used for our test user then we will associate
the DN to our Cisco IP Communicator (CIPC). When the line is busy or there is no answer
calls will be sent to voicemail (at DN 2000) sent across our SIP trunk to our CUE router.

ephone-dn 24
number 199
label testuser - 199
name testuser

ephone 24
username "testuser"
mac-address 0001.4A25.68E0
type CIPC
button 1:24

STEP 5: VOICEMAIL PILOT ON CME
We will configure the voicemail pilot number under our CME telephony-service to be 2000
matching our configured VOIP dial peer. This creates a voicemail speed dial once the
voicemail button is pressed on the IP Phone.

telephony-service
voicemail 2000

STEP 6: AA EXAMPLE USE
Configuring AA is the same as in any case. In our case any call received on FXO port 0/0/0
will be forwarded to AA pilot number 2001 (matching a configured VOIP dial peer) sent
across our SIP trunk to our CUE router.

voice-port 0/0/0
pre-dial-delay 0
no vad
caller-id enable

STEP A: SUMMARY CONFIGURATION

voice service voip

codec g711ulaw
no vad

incoming called-number 80[0,1]...
codec g711ulaw

telephony-service
voicemail 2000

ephone-dn 20
number 800...
mwi on

ephone-dn 21
number 801...
mwi off

voice-port 0/0/0
pre-dial-delay 0
no vad
caller-id enable

ephone-dn 24
number 199
label testuser - 199
name testuser

ephone 24
username "testuser"
mac-address 0001.4A25.68E0
type CIPC
button 1:24

5.7.2 CUE Router (Cisco CUE Router Configuration)

STEP 1: VOIP DIAL PEER TO CISCO CUE ROUTER
Next lets configure a SIP trunk matching 1XX directory numbers (any DN from 100 to 199)
pointing to the CME router (at 192.168.3.254).

destination-pattern 1..
codec g711ulaw
no vad

STEP 2: VOIP DIAL PEER TO CUE
Next lets configure a SIP trunk matching 2XXX directory numbers (any DN from 2000 to
2999) pointing to the CUE service engine where our CUE services exist such as voicemail.

codec g711ulaw
no vad

STEP 3: VOIP DIAL PEER FOR MWI
When a new voicemail message is received for one of the subscribers configured on the
CUE service engine we will want to forward the MWI messages across our SIP trunk to the
CME router (at 192.168.3.254) where our IP phones are registered at.

destination-pattern 80[0,1]...
codec g711ulaw
no vad

5.7.3 CUE Router (CUE Configuration)

STEP 1: SIP TRUNK CONFIGURATION
Configure the SIP trunk to the local Cisco router where the CUE service engine is installed by
specifying the default gateway (the IP on the local router).

ccn subsystem sip
end subsystem

STEP 2: DNS FOR VM, AA, AND PM
Next we will configure three call handlers (or triggers) for each CUE service; VM, AA, and
PM.

enabled
maxsessions 6
end trigger

enabled
locale "en_US"
maxsessions 6
end trigger

enabled
idletimeout 5000
locale "en_US"
maxsessions 1
end trigger

STEP 3: MWI CONFIGURATION
Next we will configure the MWI application specifying the MWI ON and OFF directory
numbers.

enabled
maxsessions 4
script "setmwi.aef"
end application


STEP 4: CREATE MAILBOXES
Next we will create our user mailbox account where our test username will be U199 with DN
199.


username U199 create
username U199 phonenumber "199"

voicemail mailbox owner "U199" size 420
description "Test User"
end mailbox

NOTE: the following is configured in the enable mode and NOT the config mode.

username U199 fullname first Test last User display "Test User" password
cisco6778

STEP A: SUMMARY CONFIGURATION

destination-pattern 1..
codec g711ulaw
no vad

codec g711ulaw
no vad

destination-pattern 80[0,1]...
codec g711ulaw
no vad

--------------------------

username U199 create
username U199 phonenumber "199"


enabled
maxsessions 4
script "setmwi.aef"
end application


ccn subsystem sip
end subsystem

enabled
maxsessions 6
end trigger

enabled
locale "en_US"
maxsessions 6
end trigger

enabled
idletimeout 5000
locale "en_US"
maxsessions 1
end trigger

voicemail mailbox owner "U199" size 420
description "Test User"
end mailbox

username U199 fullname first Test last User display "Test User" password
cisco6778

5.8 Live Record

Live Record is a feature that is configured within CME and CUE. Live Record allows users to
record calls during a connected phone conversation by pressing the Live Record softkey on
their phone. These recorded messages are left on the subscribers voice mailbox as a
voicemail. During the recording, periodical beeps will occur during the call, which is required
for certain national legislations like Australia. Ad-Hoc Conferencing is required to be
configured and working in order for Live Record to work.

NOTE: Make sure to upgrade CME to version 7.x or higher before configuring Live
Record, which is what our configuration is based upon. Those instructions are
shown in this workbook.

STEP 1: PHONE TEMPLATE (ON CME)
First we need to configure or redefine our ephone template to include the LiveRcd softkey
for Live Record. This would be configured under the connected softkey profile since Live
Record happens during a connected call. Once the softkey template is configured we would
apply that to all phones on our network that will use this.

ephone-template 1
softkeys connected LiveRcd Confrn Hold Park Trnsfer TrnsfVM

ephone 2
ephone-template 1

ephone 6
ephone-template 1

NOTE: These are for two Cisco 7970 phones.

STEP 2: LIVE RECORD CONFIGURATION ON CME
To configure Live Record, two things are required for us to know. First we need to know
what our voicemail pilot number is and second is to know what our Live Record pilot number
will be. In our configuration, our VM pilot number is 6000 and we will configure our Live
Record pilot number to be 6005.

Basically what we will need to do is enable Live Record under our CME telephony service
specifying what the Live Record number will be, in our case that would be 6005. If the
voicemail pilot number has not been configured we will also add that to our CME telephony
service configuration.

Any voicemail calls or use of Live Record (to DN 6005) will automatically be forwarded to the
voicemail pilot number of 6000 where a dial peer is already configured pointing any
unmatched DN in the range of 6000 to 6999 to the CUE service engine.

telephony-service
live-record 6005
voicemail 6000

ephone-dn 16
number 6005

codec g711ulaw
no vad

Once our CME configuration portion of Live Record is completed we need to reset each
phone where our update phone soft-key template was applied.

ephone 2
reset

ephone 6
reset

STEP 3: AD-HOC CONFERENCING ON CME
As we discussed before Ad-Hoc conferencing is required for Live Record to work because it
is basically creating a conference call where both participants are added to a single call
automatically then the call is recorded.

This configuration can be found under Configuration for CME > Conferencing > Ad-hoc
Conferencing for completing that configuration required.

STEP 4: LIVE RECORD CONFIGURATION ON CUE
The configuration on CUE for Live Record is a lot simpler than what is needed on CME. We
will specify the Live Record pilot number and the beep duration (configured in milliseconds)
when Live Record is active, so the caller is aware that the conversation is being recorded.

voicemail live-record beep duration 1000
voicemail live-record pilot-number 6005

STEP 5: USING LIVE RECORD
Using Live Record is very simple. When you have a connected call (placed or received) you
will likely inform the caller that you will record the conversation. Next, simply press the
LiveRcd softkey on the phone and that will create an Ad-hoc conference bridge joining both
callers recording the conversation hearing a periodic beep during the call.

Once the call is finished we can simply end the call and a new voicemail message will be
waiting with the conversation we just recorded.

5.9 Downgrade CUE software

Live Record is a feature that is configured within CME and CUE. Live Record allows users to
record calls during a connected phone conversation by pressing the Live Record softkey on
their phone. These recorded messages are left on the subscribers voice mailbox as a
voicemail. During the recording, periodical beeps will occur during the call, which is required
for certain national legislations like Australia. Ad-Hoc Conferencing is required to be
configured and working in order for Live Record to work.

STEP 1: OVERVIEW
Why downgrade? Well there are many reasons why downgrading the CUE software version
is important due to software related bugs or maybe licensing limitations. We completed a
CUE downgrade recently for a client from 7.1 back to 7.0.1.

CUE 7.1 introduces a new licensing model that isn't fine-tuned and as a result we needed to
downgrade back to CUE 7.0.1, which works great and provide many features that are
important for our clients.

For example, on our CUE device, when we look at the licensing we see that voicemail is
disabled and that there are no mailbox licenses available. Therefore, doing this upgrade has
brought down our voicemail services for the client. Therefore, we need to downgrade back to
a working environment.

rhg-cue01-sf-ca# show license status application
voicemail disabled, no activated mailbox license available
ivr disabled, no activated ivr session license available

STEP 2: DOWNLOAD CUE SOFTWARE FROM CISCO SOFTWARE CENTER
We need to make sure we have all the files required for CUE 7.0.1.

Reference section 5.3 for the process of downloading the right files are needed, setting up
the FTP server, and other steps.

STEP 3: CUE SOFTWARE DOWNLOAD TO FLASH
With our software on our FTP server we will do a "clean" download of our CUE 7.0.1
package. This will erase the current software on the CUE module and do a fresh "clean"
install of another CUE software image. In our case CUE version 7.0.1.

It's also a very good practice to copy any configuration on the CUE to your computer and go
through any voicemail messages that may exist on the CUE system before we do a
downgrade. A warning message is seen when we start the software download process.

When we execute the below command it will start downloading the necessary files for CUE
7.0.1 from the FTP server including the language packs that need to be downloaded and
various CUE scripts will run.

rhg-cue01-sf-ca# software download clean cue-vm-k9.nme.7.0.1.pkg


WARNING:: This command will download the necessary software to
WARNING:: complete a clean install. It is recommended that a backup be done

Would you like to continue?[confirm]

Downloading ftp cue-vm-k9.nme.7.0.1.pkg

- Parsing package manifest files... complete.
Validating installed manifests ............complete.
- Checking Package dependencies... complete.

Downloading ftp cue-vm-langpack.nme.7.0.1.pkg

Found Add-On Subsystem SID: e2e81cc6-39b5-47e1-9f83-b83c897fc50c Name: CUE
Voicemail Language Support Version: 7.0.1.0
....

- Checking Manifest dependencies for subsystems in the install candidate
list...
complete
Starting payload download
File : cue-vm-full-k9.nme.7.0.1.prt1 Bytes : 89894237
Validating payloads match registered checksums...
- cue-vm-full-k9.nme.7.0.1.prt1
............................................................................
..........verified
Extracting install scripts ...
starting_phase:
install_files.sh /dwnld/.script_work_order
add_file /dwnld/pkgdata/cue-vm-full-k9.nme.7.0.1.prt1 15
/dwnld/scripts/e2e81cc6-39b5-47e1-9f83-b83c897fc50c
usr/bin/products/cue/lang_ui_script.py tgz
Scripts extraction complete.
Remove scripts work order /dwnld/.script_work_order
Running Script Processor for ui_install

Maximum 5 language add-ons allowed for this platform.
Please select language(s) to install from the following list:

Language Installation Menu:

# Selected SKU Language Name (version)
----------------------------------------------------------------------
1 ITA CUE Voicemail Italian (7.0.1.0)
2 ESP CUE Voicemail European Spanish (7.0.1.0)
3 ENU CUE Voicemail US English (7.0.1.0)
4 FRA CUE Voicemail European French (7.0.1.0)
5 ESO CUE Voicemail Latin American Spanish (7.0.1.0)
6 ESM CUE Voicemail Mexican Spanish (7.0.1.0)
7 ARA CUE Voicemail Arabic (7.0.1.0)
8 NLD CUE Voicemail Dutch (7.0.1.0)
9 SVE CUE Voicemail Swedish (7.0.1.0)
10 NOR CUE Voicemail Norwegian (7.0.1.0)
11 FRC CUE Voicemail Canadian French (7.0.1.0)
12 TUR CUE Voicemail Turkish (7.0.1.0)
13 HUN CUE Voicemail Hungarian (7.0.1.0)

14 ENG CUE Voicemail UK English (7.0.1.0)
15 DEU CUE Voicemail German (7.0.1.0)
16 DAN CUE Voicemail Danish (7.0.1.0)
17 PTB CUE Voicemail Brazilian Portuguese (7.0.1.0)
18 KOR CUE Voicemail Korean (7.0.1.0)
19 CHS CUE Voicemail Mandarin Chinese (7.0.1.0)
20 JPN CUE Voicemail Japanese (7.0.1.0)
21 RUS CUE Voicemail Russian (7.0.1.0)
----------------------------------------------------------------------

Available commands are:
# - enter the number for the language to select one
r # - remove the language for given #
i # - more information about the language for given #
x - Done with language selection

Enter Command:3


----------------------------------------------------------------------
3 * ENU CUE Voicemail US English (7.0.1.0)
----------------------------------------------------------------------


Enter Command:x
ui_install scripts executed successfully.

STEP 4: CONFIRM CUE SOFTWARE DOWNLOAD
Before we continue let's confirm if the CUE download process has completed successfully.

rhg-cue01-sf-ca# software download status
Download request completed successfully.
rhg-cue01-sf-ca#

STEP 5: INSTALL CUE SOFTWARE (CLEAN)
Now we will installed the downloaded CUE software on our CUE module erasing the current
CUE image and installing a clean CUE image

Again make sure that your configuration is backed up before you continue. After the new OS
is installed the CUE system will then reboot as part of its process.

wcm-cue01-sf-ca# software install clean cue-vm-k9.nme.7.0.1.pkg

WARNING:: This command will install the necessary software to
WARNING:: complete a clean install. It is recommended that a backup be done

Would you like to continue?[confirm]
Validating installed manifests ............complete.
Found Add-On Subsystem SID: e2e81cc6-39b5-47e1-9f83-b83c897fc50c Name: CUE
Voicemail Language Support Version: 7.0.1.0
....

Maximum 5 language add-ons allowed for this platform.
Please select language(s) to install from the following list:


----------------------------------------------------------------------
3 ENU CUE Voicemail US English (7.0.1.0)

----------------------------------------------------------------------


Enter Command:3


----------------------------------------------------------------------
3 * ENU CUE Voicemail US English (7.0.1.0)
----------------------------------------------------------------------


Enter Command:x
ui_install scripts executed successfully.
Downloading payload(s) complete
Validating payloads match registered checksums...
- cue-vm-en_US-langpack.nme.7.0.1.prt1 ...........................verified
The system will be brought to offline state for a brief period
and will be brought back to online state automatically

STEP 6: CUE RESTARTS
We now see the CUE rebooting after completing its software installation.

System Now Booting ...

704 832 968 1040 1172 1184 1196 1208 1220 1228 1240 1260 1276 1288 1304 1320
1332 1348 1368 1380 1644 1784 2060 2204 2344 2860 3376 3640 3904 4168 RSA
decrypt returned:33
bd4359c1c1a02f7dd52512f52e502438

Booting from Secure secondary boot loader..., please wait.

[BOOT-ASM]

Updating flash with bootloader configuration: 1
Please wait ...
.............done.
so on ....

STEP 7: RESTORE CONFIGURATION OR SETUP WIZARD
Since we did a "clean" install it will ask if we want to enter a new configuration or restore our
working configuration stored in the CUE flash. If a configuration is found in the flash it will
ask us if we want to restore the configuration from the flash. If no configuration is found it will
start the setup wizard instead by configuring a base configuration.

IMPORTANT::
IMPORTANT:: Welcome to Cisco Systems Service Engine
IMPORTANT:: post installation configuration tool.
IMPORTANT::
IMPORTANT:: This is a one time process which will guide
IMPORTANT:: you through initial setup of your Service Engine.
IMPORTANT:: Once run, this process will have configured
IMPORTANT:: the system for your location.
IMPORTANT::
IMPORTANT:: If you do not wish to continue, the system will be halted
IMPORTANT:: so it can be safely removed from the router.
IMPORTANT::

Do you wish to start configuration now (y,n)? y
Are you sure (y,n)? y

IMPORTANT::
IMPORTANT:: A configuration has been found in flash. You can choose
IMPORTANT:: to restore this configuration into the current image.
IMPORTANT::
IMPORTANT:: A stored configuration contains some of the data from a
IMPORTANT:: previous installation, but not as much as a backup.
IMPORTANT::
IMPORTANT:: If you are recovering from a disaster and do not have a
IMPORTANT:: backup, you can restore the saved configuration.
IMPORTANT::
IMPORTANT:: If you choose not to restore the saved configuration, it
IMPORTANT:: will be erased from flash.
IMPORTANT::


Would you like to restore the saved configuration? (y,n)

STEP 8: CUE SYSTEM ON-LINE
Once the configuration is restored or the setup wizard completes it will then bring the CUE
system ONLINE

SYSTEM ONLINE

5.10 Basic CUE Start-Up Wizard

For a new CUE system or if the CUE cannot locate a configuration file in the flash it will start
the setup wizard similar to the setup wizard for Cisco IOS routers and switches.

Keep in mind that this is only a base configuration and does not provide any configuration for
features related to voicemail or auto attendant.

Below is what the setup wizard will configure:
Hostname
Domain Name
DNS Server (Primary, Secondary)
NTP Server (Primary, Secondary)
Time Zone
Admin account

Once the wizard completes the system will come ONLINE. If we cancel the setup wizard it
will halt and put the CUE system OFFLINE.

Below is an example when we run the setup wizard on our CUE:

STEP 1: HOSTNAME
Enter Hostname
(my-hostname, or enter to use se-10-67-79-2): rhg-cue01-sf-ca

STEP 2: DOMAIN NAME
Enter Domain Name
(mydomain.com, or enter to use localdomain): routehub.com

STEP 3: DNS
IMPORTANT:: DNS Configuration:
IMPORTANT::
IMPORTANT:: This allows the entry of hostnames, for example foo.cisco.com,
instead
IMPORTANT:: of IP addresses like 1.100.10.205 for application configuration.
In order
IMPORTANT:: to set up DNS you must know the IP address of at least one of
your
IMPORTANT:: DNS Servers.

Would you like to use DNS (y,n)?y

Enter IP Address of the Primary DNS Server
(IP address): 4.2.2.2
Found server 4.2.2.2

Enter IP Address of the Secondary DNS Server (other than Primary)
(IP address, or enter to bypass):


STEP 4: NTP
Enter Fully Qualified Domain Name(FQDN: e.g. myhost.mydomain.com)
or IP address of the Primary NTP server
(FQDN or IP address, or enter for 10.67.79.1):
Found server 10.67.79.1

Enter Fully Qualified Domain Name(FQDN: e.g. myhost.mydomain.com)
or IP address of the Secondary NTP Server
(FQDN or IP address, or enter to bypass):

STEP 5: TIME ZONE
Please identify a location so that time zone rules can be set correctly.
Please select a continent or ocean.
1) Africa 4) Arctic Ocean 7) Australia 10) Pacific
Ocean
2) Americas 5) Asia 8) Europe
3) Antarctica 6) Atlantic Ocean 9) Indian Ocean
#? 2
Please select a country.
1) Anguilla 27) Honduras
2) Antigua & Barbuda 28) Jamaica
3) Argentina 29) Martinique
4) Aruba 30) Mexico
5) Bahamas 31) Montserrat
6) Barbados 32) Netherlands Antilles
7) Belize 33) Nicaragua
8) Bolivia 34) Panama
9) Brazil 35) Paraguay
10) Canada 36) Peru
11) Cayman Islands 37) Puerto Rico
12) Chile 38) St Barthelemy
13) Colombia 39) St Kitts & Nevis
14) Costa Rica 40) St Lucia
15) Cuba 41) St Martin (French part)
16) Dominica 42) St Pierre & Miquelon
17) Dominican Republic 43) St Vincent
18) Ecuador 44) Suriname
19) El Salvador 45) Trinidad & Tobago
20) French Guiana 46) Turks & Caicos Is
21) Greenland 47) United States
22) Grenada 48) Uruguay
23) Guadeloupe 49) Venezuela
24) Guatemala 50) Virgin Islands (UK)
25) Guyana 51) Virgin Islands (US)
26) Haiti
#? 47
Please select one of the following time zone regions.
1) Eastern Time
2) Eastern Time - Michigan - most locations
3) Eastern Time - Kentucky - Louisville area
4) Eastern Time - Kentucky - Wayne County
5) Eastern Time - Indiana - most locations
6) Eastern Time - Indiana - Daviess, Dubois, Knox & Martin Counties
7) Eastern Time - Indiana - Starke County
8) Eastern Time - Indiana - Pulaski County
9) Eastern Time - Indiana - Crawford County
10) Eastern Time - Indiana - Switzerland County
11) Central Time
12) Central Time - Indiana - Perry County
13) Eastern Time - Indiana - Pike County

14) Central Time - Michigan - Dickinson, Gogebic, Iron & Menominee Counties
15) Central Time - North Dakota - Oliver County
16) Central Time - North Dakota - Morton County (except Mandan area)
17) Mountain Time
18) Mountain Time - south Idaho & east Oregon
19) Mountain Time - Navajo
20) Mountain Standard Time - Arizona
21) Pacific Time
22) Alaska Time
23) Alaska Time - Alaska panhandle
24) Alaska Time - Alaska panhandle neck
25) Alaska Time - west Alaska
26) Aleutian Islands
27) Hawaii
#? 21

The following information has been given:

United States
Pacific Time

Therefore TZ='America/Los_Angeles' will be used.
Is the above information OK?
1) Yes
2) No
#? 1

Local time is now: Thu Mar 18 02:35:33 PDT 2010.
Universal Time is now: Thu Mar 18 09:35:33 UTC 2010.

STEP 6: ADMIN ACCOUNT
Configuring the system. Please wait...

IMPORTANT::
IMPORTANT:: Administrator Account Creation
IMPORTANT::
IMPORTANT:: Create an administrator account. With this account,
IMPORTANT:: you can log in to the Cisco Unity Express GUI and
IMPORTANT:: run the initialization wizard.
IMPORTANT::

Enter administrator user ID:
(user ID): admin
Enter password for admin:
(password):
Confirm password for admin by reentering it:
(password):

STEP 7: CUE SYSTEM ON-LINE
SYSTEM ONLINE

6 Monitor
6.1 Operations

6.1.1 IP Phones
Below are some show commands reflecting how to monitor your IP Telephony environment
and details on how are UC500 appliance is configured for CallManager Express.

The following command shows all IP Phone models that CallManager Express knows about
and is supported by the appliance.
uc01tra#show ephone ?
7902 7902 phone status
7941GE 7941GE phone status
7961GE 7961GE phone status
H.H.H mac address
anl ANL port status
ata ata phone status
attempted-registrations Attempted ephone list
bri BRI port status
cfa registered ephones with call-forward-all set
dn Dn with tag assigned
dnd registered ephones with do-not-disturb set
login phone login status
offhook Offhook phone status
overlay registered ephones with overlay DNs
phone-load Ephone phoneload information
registered Registered ephone status
remote non-local phones (with no arp entry)
ringing Ringing phone status
sockets Active ephone sockets
summary Summary of all ephone
tapiclients Ephone status of tapi client
telephone-number Telephone number assigned
unregistered Unregistered ephone status
| Output modifiers
<cr>

The following command gives a good summary of all IP Phones that are registered, not
registered to CallManager Express including the number of active conference call sessions.
This is a good command to use for knowing what phones are registered with the UC express
server, the IP addresses assigned and what lines are mapped to that phone.
uc01tra#show ephone summary

hairpin_block:
ephone-1 Mac:001B.D52C.77C5 TCP socket:[-1] activeLine:0 DECEASED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.67.78.128 7906 keepalive 114 1:13

ephone-5 Mac:0012.00A7.72EA TCP socket:[3] activeLine:0 REGISTERED
IP:10.67.78.36 Telecaster 7960 keepalive 1645 1:10 2:12

ephone-6 Mac:0011.932B.8B15 TCP socket:[2] activeLine:1 REGISTERED
IP:10.67.78.31 7970 keepalive 3689 1:10 2:11 3:13

ephone-7 Mac:000D.288E.3F4A TCP socket:[1] activeLine:0 REGISTERED
IP:10.67.78.111 7920 keepalive 53050 1:10 2:12

Max 14, Registered 3, Unregistered 0, Deceased 1, Sockets 4
ephone_send_packet process switched 0

Max Conferences 8 with 0 active (8 allowed)
Skinny Music On Hold Status
Active MOH clients 0 (max 156), Media Clients 0, B-ACD Clients 0
File music-on-hold.au type AU Media_Payload_G711Ulaw64k 160 bytes

The previous command gave us a summary of all ephones on our voice network, the
following command shows only registered IP Phones to CallManager Express. You will
notice an active call on ephone-6. That info will look different compared to the other entries
listed. Ephone-6 will have details of Jitter, Latency, traffic, CODEC, and more. Under button
1 associated with DN 6700 the status will say CONNECTED meaning we have an active
call from this number.
uc01tra#show ephone registered

ephone-5 Mac:0012.00A7.72EA TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 6 and
Server in ver 6
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.67.78.36 50058 Telecaster 7960 keepalive 1645 max_line 6
button 1: dn 10 number 6700 CH1 CONNECTED CH2 IDLE shared
button 2: dn 12 number A5002 auto dial A5001 CH1 IDLE shared
paging-dn 1
Username: dn6700

ephone-6 Mac:0011.932B.8B15 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 6 and
Server in ver 6
IP:10.67.78.31 49949 7970 keepalive 3689 max_line 8
button 2: dn 11 number A5001 auto dial A5002 CH1 IDLE
button 3: dn 13 number 7700 CH1 IDLE CH2 IDLE
Active Call on DN 10 chan 1 :6700 10.67.78.31 18468 to 10.67.78.1 2000 via 10.67.78.31
G711Ulaw64k 160 bytes no vad
Tx Pkts 991 bytes 170452 Rx Pkts 995 bytes 171140 Lost 0
Jitter 0 Latency 8 callingDn -1 calledDn 22 (media path callID 63766 srcCallID 63768)

ephone-7 Mac:000D.288E.3F4A TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 5 and
Server in ver 5
IP:10.67.78.111 1050 7920 keepalive 53051 max_line 6

button 2: dn 12 number A5002 auto dial A5001 CH1 IDLE shared

6.1.2 Conferencing and DSP resources
Looking a bit deeper with our IP Telephony environment, we can look at details of our
conferencing and DSP resources on our UC500 appliance.

The following command shows all active conference sessions including the conference
directory number. Here we see one conference session on number 6999 is in place.
uc01tra#show ephone-dn conference
type active inactive numbers
=======================================
Meetme 1 7 6999
DN tags: 22, 23, 24, 25

Similar to the previous command.
uc01tra#show ephone-dn conference meetme
type active inactive numbers
=======================================
Meetme 1 7 6999
DN tags: 22, 23, 24, 25

uc01tra#show ephone-dn statistics
Total Calls 900
Stats may appear to be inconsistent for conference or shared line cases

DN 1 chan 1 incoming 1 answered 0 outgoing 0 answered 0 busy 0
Far-end disconnect at: connect 1 alert 0 hold 0 ring 0
Last 64 far-end disconnect cause codes
16 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0

0 19 16 28 28 28 19 16 19 19 19 28 19 19 19 19
19 19 16 19 16 16 19 16 16 16 63 63 16 1 1 16
1 1 1 1 1 28 1 19 16 63 63 38 63 63 63 63
63 38 63 19 63 63 28 28 28 28 63 16 63 19 19 19

DN 10 chan 1 (6700) voice quality statistics for current call
Call Ref 900 called 6999 calling 6700
Current Tx Pkts 3241 bytes 557452 Rx Pkts 3245 bytes 558140 Lost 0
Jitter 0 Latency 0
Worst Jitter 0 Worst Latency 11
Signal Level to phone 0 (-78 dB) peak 0 (-78 dB)
Packets counted by router 3247

0 28 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
local EndCall pressed

DN 10 chan 2 (6700) voice quality statistics for last call
Total Tx Pkts 837 bytes 143964 Rx Pkts 777 bytes 133644 Lost 22
Final Jitter 24 Latency 373 Lost 0


0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
local phone on-hook

0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0

DN 12 chan 1 (A5002) voice quality statistics for last call
Call Ref 581 called A5002 calling A5001

19 16 16 16 16 19 16 19 16 16 16 16 16 16 16 19
16 19 19 19 19 19 19 19 19 19 19 19 19 19 19 19
19 19 19 19 19 19 19 19 19 16 19 19 16 19 1 1
1 1 1 19 1 19 0 0 0 0 0 0 0 0 0 0


28 28 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
local EndCall pressed

Call Ref 868 called calling

The following shows a summary of all IP Phones and active calls. In this case we have an
active conference call using G711 CODEC. Also notice the ports, where 50/0/10 is an IP
Phone with directory number 6700 and port 50/0/22 is the meetme number of 6999.
uc01tra#show ephone-dn summary
PORT CH DN STATE MWI_STATE CODEC VAD VTSP STATE VPM STATE
======== == ======== ========= ===== === =========== =========
50/0/10 1 CONNECTED NONE g711ulaw n S_CONNECT
EFXS_CONNECT
50/0/10 2 IDLE NONE - - - EFXS_ONHOOK
50/0/22 1 CONNECTED NONE g711ulaw n S_CONNECT EFXS_CONNECT

The following command shows details of our conference bridge and all active conference
calls. It also tells us, which IP Phone/Directory number initiated (the Master) the conference
bridge. In our case it is directory number 6700. We also see the conference bridge/meetme
directory number being 6999.
uc01tra#show telephony-service conference hardware detail
Conference Type Active Max Peak Master MasterPhone Last
cur(initial)
=================================================================================
6999 Meetme 1 8 1 6700 6700 6 ( 6) 6700 6700
Conference parties:
6700 6700

The following command shows hardware specifics of our DSP resources, which is required
for conferencing to work.
uc01tra#show voice dsp

DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT
==== === == ======== ========== ===== ======= === == ========= == ===== ============
edsp 001 01 g711ulaw 0.1 IDLE 50/0/1.1
edsp 002 01 g729r8 p 0.1 IDLE 50/0/2.1
edsp 003 01 g711ulaw 0.1 IDLE 50/0/10.1
edsp 004 02 g711ulaw 0.1 IDLE 50/0/10.2
edsp 005 01 g711ulaw 0.1 IDLE 50/0/11.1
edsp 006 01 g711ulaw 0.1 IDLE 50/0/12.1
edsp 007 01 g729r8 0.1 IDLE 50/0/13.1
edsp 008 02 g711ulaw 0.1 IDLE 50/0/13.2
edsp 009 01 g729r8 p 0.1 IDLE 50/0/20.1
edsp 010 01 g729r8 p 0.1 IDLE 50/0/21.1
edsp 011 01 g711ulaw 0.1 IDLE 50/0/22.1
edsp 012 02 g711ulaw 0.1 IDLE 50/0/22.2
edsp 013 01 g729r8 p 0.1 IDLE 50/0/23.1
edsp 014 02 g729r8 p 0.1 IDLE 50/0/23.2
edsp 015 01 g729r8 p 0.1 IDLE 50/0/24.1
edsp 016 02 g729r8 p 0.1 IDLE 50/0/24.2
edsp 017 01 g729r8 p 0.1 IDLE 50/0/25.1
edsp 018 02 g729r8 p 0.1 IDLE 50/0/25.2

----------------------------FLEX VOICE CARD 0 ------------------------------
*DSP VOICE CHANNELS*

CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND : (bad)bad (shut)shutdown (dpend)download pending

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ========= ========== ===== ======= === == ========= == ==== ============
*DSP SIGNALING CHANNELS*
===== === == ========= ========== ===== ======= === == ========= == ==== ============
C5510 001 01 {flex} 21.4.0 alloc idle 0 0 0/0/0 02 0 46/0
C5510 001 02 {flex} 21.4.0 alloc idle 0 0 0/0/1 02 0 46/0
C5510 001 03 {flex} 21.4.0 alloc idle 0 0 0/0/2 06 0 46/0
C5510 001 04 {flex} 21.4.0 alloc idle 0 0 0/0/3 06 0 46/0
C5510 001 05 {flex} 21.4.0 alloc idle 0 0 0/1/0 02 0 2425/0
C5510 001 06 {flex} 21.4.0 alloc idle 0 0 0/1/1 06 0 36/0
C5510 001 07 {flex} 21.4.0 alloc idle 0 0 0/1/2 10 0 36/0
C5510 001 08 {flex} 21.4.0 alloc idle 0 0 0/1/3 14 0 36/0
C5510 001 09 {flex} 21.4.0 alloc idle 0 0 0/4/0 02 0 0/0
C5510 001 10 {flex} 21.4.0 alloc idle 0 0 0/4/1 02 0 0/0
------------------------END OF FLEX VOICE CARD 0 ----------------------------

uc01tra#show voice dsp voice

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT
==== === == ======== ========== ===== ======= === == ========= == ===== ============
edsp 001 01 g711ulaw 0.1 IDLE 50/0/1.1
edsp 002 01 g729r8 p 0.1 IDLE 50/0/2.1
edsp 003 01 g711ulaw 0.1 IDLE 50/0/10.1
edsp 004 02 g711ulaw 0.1 IDLE 50/0/10.2
edsp 005 01 g711ulaw 0.1 IDLE 50/0/11.1
edsp 006 01 g711ulaw 0.1 IDLE 50/0/12.1
edsp 007 01 g729r8 0.1 IDLE 50/0/13.1
edsp 008 02 g711ulaw 0.1 IDLE 50/0/13.2
edsp 009 01 g729r8 p 0.1 IDLE 50/0/20.1
edsp 010 01 g729r8 p 0.1 IDLE 50/0/21.1
edsp 011 01 g711ulaw 0.1 IDLE 50/0/22.1
edsp 012 02 g711ulaw 0.1 IDLE 50/0/22.2
edsp 013 01 g729r8 p 0.1 IDLE 50/0/23.1
edsp 014 02 g729r8 p 0.1 IDLE 50/0/23.2
edsp 015 01 g729r8 p 0.1 IDLE 50/0/24.1
edsp 016 02 g729r8 p 0.1 IDLE 50/0/24.2
edsp 017 01 g729r8 p 0.1 IDLE 50/0/25.1
edsp 018 02 g729r8 p 0.1 IDLE 50/0/25.2

----------------------------FLEX VOICE CARD 0 ------------------------------
*DSP VOICE CHANNELS*

CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND : (bad)bad (shut)shutdown (dpend)download pending

===== === == ========= ========== ===== ======= === == ========= == ==== ============
C5510 001 01 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 02 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 03 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 04 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 05 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 06 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 07 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 08 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 09 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 10 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 11 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 12 None 21.4.0 idle idle 0 0 0 0/0

C5510 001 13 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 14 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 15 None 21.4.0 idle idle 0 0 0 0/0
C5510 001 16 None 21.4.0 idle idle 0 0 0 0/0
C5510 002 01 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 02 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 03 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 04 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 05 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 06 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 07 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 08 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 09 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 10 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 11 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 12 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 13 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 14 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 15 None 1.1.137 idle idle 0 0 0 0/0
C5510 002 16 None 1.1.137 idle idle 0 0 0 0/0
------------------------END OF FLEX VOICE CARD 0 ----------------------------

The following command gives DSP resource information on the UC500 and how it is used. In
our case, DSP resources are being used for CONFERENCING and what is most important to
confirm is if the admin and operation state are both UP and ACTIVE. The command will also
provide information on the number of DSP resources configured and available for other
sources.
uc01tra#show dspfarm all
Dspfarm Profile Configuration

Profile ID = 1, Service = CONFERENCING, Resource ID = 1
Profile Description :
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP Status : ASSOCIATED
Resource Provider : FLEX_DSPRM Status : UP
Number of Resource Configured : 2
Number of Resource Available : 2
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required
Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required
Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required
Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required
Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required
Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 2 1.1.137 UP N/A FREE conf 1 - - -
0 2 1.1.137 UP N/A FREE conf 1 - - -

Total number of DSPFARM DSP channel(s) 2


6.1.3 Dial Plan and Cisco CallManager Express
This sections supplies additional information and details on our CallManager Express setup.

The following command shows our dial plan or route pattern table and how calls are routed.
For example, anyone on our IPT network who dials 7700 will be sent to port 50/0/13, which is
associated with our 7970 IP Phone. There are other destination ports such as for SIP trunks
and our FXO Port (0/1/0), which is connected to our PSTN. If we dial 9 first then a full
number it would match destination pattern 9.T and route the call through port 0/1/0, which is
the FXO port. We also see that this is a pots port and not a voip port, which is true
because we have an analog line connected to the FXO port.
uc01tra#show dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
100 pots up up 9.T 0 up 0/1/0
600 voip up up 6000 0 syst ipv4:192.168.5.2
20001 pots up up 6700$ 0 50/0/10
20002 pots up up 8000.... 0 50/0/20
20003 pots up up 8001.... 0 50/0/21
20004 pots up up A5001$ 0 50/0/11
20005 pots up up A5002$ 0 50/0/12
20006 pots up up 7700$ 0 50/0/13
20007 pots up up 19252302203$ 9 50/0/13
11 voip up up 8[2-9]..[2-9]...- 0 syst sip-server
...
12 voip up up 8[0-1][2-9]..[2-- 0 syst sip-server
9]......
13 voip up down 8911 0 syst
601 voip up up 4106... 0 syst ipv4:192.168.0.250
20008 pots up up 6001$ 0 50/0/1
20009 pots up up 6002$ 0 50/0/2
20010 pots up up 6999$ 0 50/0/22
20011 pots up up 6999$ 1 50/0/23
20012 pots up up 6999$ 2 50/0/24
20013 pots up up 6999$ 3 50/0/25

This command is very useful and information to know how CallManager Express is
configured. This can tell us the version of CallManager Express installed and the details on
how it is configured such as the source IP used for CallManager Express to the IP Phone
services used for the IP Phones. Our CallManager Express version on the UC500 is 4.2.
uc01tra#show telephony-service
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home
.html

ip qos dscp:
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for serviceservice directed-pickup
load 7914 S00104000100
load 7902 CP7902080001SCCP051117A
load 7906 SCCP11.8-0-3S
load 7911 SCCP11.8-0-3S
load 7921 CP7921G-1.0.1
load 7936 cmterm_7936.3-3-5-0
load 7960-7940 P0030702T023
load 7941 TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S

load 7971 TERM70.7-0-3-0S
max-ephones 14
max-dn 56
dspfarm units 3
dspfarm transcode sessions 0
dspfarm 1 mtp001d4567c690
dspfarm 2
dspfarm 3
conference hardware
privacy
no privacy-on-hold
# of hunt-group collect data: 1
hunt-group report delay 1 hours
Number of hunt-group configured: 1
hunt-group logout DND
max-redirect 5
voicemail 6000
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
time-format 12
date-format mm-dd-yy
timezone 5 Pacific Standard/Daylight Time
url services http://10.67.78.2/voiceview/common/login.do
keepalive 30 auxiliary 30
timeout interdigit 5
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
web admin system name admin secret 5 $1$ucO7$XfKgADX7L1nzz11jbTDa./
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
Log (table parameters):
max-size: 150
retain-timer: 15
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.

This command shows all voice ports and associated directory numbers on our IPT system.
For example, we see that voice port 50/0/13 is associated with directory number 7700.
Timeout ringing means the amount of time a call is not answered before the call goes to
voicemail. Timeout interdigit means the amount of time to setup and place a call.
uc01tra#show telephony-service voice-port

voice-port 50/0/10
station-id number 6700
station-id name 6700
timeout ringing 15
!
voice-port 50/0/20
!
voice-port 50/0/21
!
voice-port 50/0/11
station-id number A5001
station-id name Intercom
!
voice-port 50/0/12
station-id number A5002
station-id name Intercom
!
voice-port 50/0/13
station-id name 7700
timeout ringing 15
!
voice-port 50/0/1
station-id name Routehub Paging S
!
voice-port 50/0/2
station-id name Routehub Call Par
!
voice-port 50/0/22
!
voice-port 50/0/23
!
voice-port 50/0/24
!
voice-port 50/0/25

On our IPT network we have enabled a feature called FAC (Forced Authorized Code), which
is different from how FAC is used with the CallManager Enterprise editions. With
CallManager Express FAC is used as short-cuts used on an IP Phone. For example, if I
press * * 9 on our IP Phone it will dial directly to voicemail.
uc01tra#show telephony-service fac

telephony-service fac standard
callfwd all **1
callfwd cancel **2
pickup local **3
pickup group **4
pickup direct **5
park **6
dnd **7
redial **8
voicemail **9
ephone-hunt join *3
ephone-hunt cancel #3
ephone-hunt hlog *4
ephone-hunt hlog-phone *5

This command shows details of our dial plan to include the directory number,
voicemail/forwarding number if there is no answer, and the associated voice port. Voice
ports labeled as 50/0/x are often referred as EFXS or IP enabled Phone ports.
uc01tra#show telephony-service dial-peer

destination-pattern 6700$
no e164 registration
huntstop
call-forward noan 6000
progress_ind setup enable 3
port 50/0/10

destination-pattern 8000....
huntstop
port 50/0/20

huntstop
port 50/0/21

destination-pattern A5001$
huntstop
port 50/0/11

destination-pattern A5002$
huntstop
port 50/0/12

huntstop
port 50/0/13

preference 9
huntstop

port 50/0/13

huntstop
port 50/0/1

huntstop
port 50/0/2

port 50/0/22

preference 1
port 50/0/23

preference 2
port 50/0/24

preference 3
port 50/0/25

Our UC500 has four FXO ports (used for connecting to a PSTN provider) and four FXS ports
(used for connecting analog phones). FXO port 0/1/0 is only being used on our appliance for
external calling to/from our PSTN.
uc01tra#show voice port

.......

Foreign Exchange Office 0/1/0 Slot is 0, Sub-unit is 1, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 64 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms

Connection Mode is plar
Connection Number is 6700
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 2 s
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 2 s
Companding Type is u-law
Region Tone is set for US

Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None

Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming):
Translation profile (Outgoing):

Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is dualtone pre-connect
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 60 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms

.......

6.1.4 SIP
This section will cover some useful commands for monitoring and looking at our working SIP
environment. Our SIP environment is setup with two functions. One, our UC500 is
configured as a SIP user agent with a SIP trunk to our SIP provider. Plus, our UC500 is
configured to act as a SIP proxy server to accept SIP client endpoints and calling.

This will show all possible commands under the sip-ua command. UA which stands for
user agent.
uc01tra#show sip-ua ?
calls Display Active SIP Calls
connections Display SIP Connections
map Display SIP status code to PSTN cause mapping table & vice versa
min-se Display Min-SE value
mwi Display SIP MWI server info
register Display SIP Register status
retry Display SIP Protocol Retry Counts

service Display SIP submode Shutdown status
statistics Display SIP UA Statistics
status Display SIP UA Listener Status
timers Display SIP Protocol Timers

This command will show us all active SIP connections to our SIP provider. As you will see,
number 19252302203 has been registered with our SIP provider, which has been configured
under sip-ua.
uc01tra#show sip-ua register status
Line peer expires(sec) registered
============ ============= ============ ===========
19252302203 20007 1299 yes
6001 20008 105 no
6002 20009 132 no
6999 20010 141 no
7700 20006 105 no
8778 40001 105 no
9.* 100 159 no

Here we have an active SIP call to a cell phone number through our SIP trunk. Here you will
see the Calling number (the source of the call) and the Called Number (the number that was
called). We also see who are SIP provider is, DTMF, codec information, and more.
SIP UAC CALL INFO

Call 1
SIP Call ID : 63689E90-D3C811DC-A4018427-
9F89A9F4@sipprovider.RouteHub.com
Bit Flags : 0xC04018 0x100 0x0
CC Call ID : 63803
Destn SIP Req Addr:Port : 10.10.10.146:5060
Destn SIP Resp Addr:Port: 10.10.10.146:5060
Destination Name : sipprovider.RouteHub.com
Media Stream 1
Negotiated Codec : g729r8 (20 bytes)
Negotiated Dtmf-relay : rtp-nte
Media Source IP Addr:Port: 6.7.7.73:16628
Media Dest IP Addr:Port : 10.10.10.146:12114
Orig Media Dest IP Addr:Port : 0.0.0.0:0


SIP UAS CALL INFO


SIP can be configured to be supported over either TCP or UDP. This command will show the
number of active SIP connections including any failures. Very useful for quick
troubleshooting.
uc01tra#show sip-ua connections udp brief

Total active connections : 2
No. of send failures : 0
No. of remote closures : 0
No. of conn. failures : 0
No. of inactive conn. ageouts : 201

We can also get details of all active SIP connection if you are unsure what the established
connections are.
uc01tra#show sip-ua connections udp detail
Total active connections : 2
No. of send failures : 0
No. of remote closures : 0
No. of conn. failures : 0
No. of inactive conn. ageouts : 201

---------Printing Detailed Connection Report---------
Note:
** Tuples with no matching socket entry
- Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port>'
to overcome this error condition
++ Tuples with mismatched address/port entry
- Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port> id <connid>'
to overcome this error condition

Remote-Agent:192.168.5.2, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size
=========== ======= =========== ===========
5060 2 Established 0

Remote-Agent:10.10.10.146, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size
=========== ======= =========== ===========
5060 1 Established 0

This command reflects SIP timer information, some that we actually configured on our UC500
under sip-ua.
uc01tra#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 100, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
tcp/udp aging 5 minutes

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED

SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl

We can also view SIP retries
uc01tra#show sip-ua retry
SIP UA Retry Values
invite retry count = 2 response retry count = 6
bye retry count = 10 cancel retry count = 10
prack retry count = 10 update retry count = 6
reliable 1xx count = 6 notify retry count = 10
refer retry count = 10 register retry count = 10
info retry count = 6 subscribe retry count = 6
options retry count = 6

This command could be a little unnecessary to show you, but I have seen setups where this
command is very necessary. Use this command to validate that the SIP UA service that has
been configured on the router is UP.
uc01tra#show sip-ua service
SIP Service is up

In our IP telephony environment we have configured numerous translation patterns a lot
pertaining to our SIP trunks. This command will show a summary of the configured
translation profiles.
uc01tra#show voice translation-profile
Translation Profile: RouteHub-tp-reject
Rule for Calling number:
Rule for Called number: 900
Rule for Redirect number:
Rule for Redirect-target number:

Translation Profile: RouteHub-tp-sip-outgoing
Rule for Calling number: 3

Translation Profile: RouteHub-tp-ucm6
Rule for Calling number:

As we stated before our UC500 is configured as a SIP proxy server. Below shows one of our
SIP profiles (for a SIP endpoint) that we configured for directory number 8701. You will see
the associated MAC address that will use this profile. This is what we see before our SIP
endpoint becomes registered to the UC500.
uc01tra#show voice register pool 1
Pool Tag 1
Config:
Mac address is 000C.F179.1682
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
keep-conference is enabled
service-control mechanism is not supported
registration Call ID is MzNlZTc4YzU0NmI1MzdhOTdjODlhNTZkN2YwNzk4MDM.
active primary line is: 8701

contact IP address: 10.67.78.101 port 2560

Dialpeers created:

Statistics:

Active registrations : 0

Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 1
Registration success : 1
Registration failed : 0
unRegister requests : 1
unRegister success : 1
unRegister failed : 0

This reflects another SIP profile configured before the SIP endpoint is registered, but for
number 8778.
Pool Tag 2
Config:
Mac address is 0019.D111.D2E8
DnD is disabled
registration Call ID is d3ac1ab6-c5fa-1810-9921-0019d111d2e8@mat-dtop01tra


Dialpeers created:

Statistics:


Now, this reflects what our SIP endpoint (configured to use number 8778) looks like when it
is registered to our UC500 SIP proxy server.
Pool Tag 2
Config:
DnD is disabled
registration Call ID is 789d1de7-c8fa-1810-90c4-0019d111d2e8@mat-dtop01tra


Dialpeers created:


session target ipv4:10.67.78.8:5061
codec g711ulaw bytes 160
after-hours-exempt FALSE

Statistics:


This command shows the details of the configuration, defaults and all, for our SIP proxy
server including our SIP profiles.
uc01tra#show voice register all
VOICE REGISTER GLOBAL
=====================
CONFIG [Version=4.2(0)]
========================
Version 4.2(0)
Mode is cme
Max-pool is 12
Max-dn is 12
Source-address is 10.67.78.1 port 5060
Time-format is 24
Date-format is YY-M-D
Time-zone is 47
Hold-alert is disabled
Mwi stutter is disabled
Mwi registration for full E.164 is disabled
Forwarding local is enabled
Dst auto adjust is enabled
start at Oct week 8 day Sun time 02:00
stop at Mar week 8 day Sun time 02:00
Max redirect number is 5
Telnet Level: 0
Tftp path is system:/cme/sipphone
Generate text file is disabled
Tftp files are not created
OS79XX.TXT is not created
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US

VOICE REGISTER DN
=================
Dn Tag 1
Config:
Number is 8701
Preference is 0
Huntstop is disabled
Name RouteHub SIP client (X-lite)
Auto answer is disabled
Dn Tag 2

Config:
Number is 8778
Preference is 0
Huntstop is disabled
Name Michel Thomatis (SIP)
Auto answer is disabled

VOICE REGISTER TEMPLATE
=======================

VOICE REGISTER DIALPLAN
=======================

VOICE REGISTER POOL
===================
Pool Tag 1
Config:
Mac address is 000C.F179.1682
DnD is disabled
registration Call ID is MzNlZTc4YzU0NmI1MzdhOTdjODlhNTZkN2YwNzk4MDM.


Dialpeers created:

Statistics:


Pool Tag 2
Config:
DnD is disabled
registration Call ID is 789d1de7-c8fa-1810-90c4-0019d111d2e8@mat-dtop01tra


Dialpeers created:

session target ipv4:10.67.78.8:5061
codec g711ulaw bytes 160
after-hours-exempt FALSE


Statistics:


This is another useful command for viewing SIP statistics on SIP registrations, requests,
failures, and much more. This is a very useful command for troubleshooting and confirming if
SIP communication is occurring properly.
uc01tra#show voice register statistics
Global statistics


Register pool 1 statistics


Register pool 2 statistics


This command shows details of an active SIP call from our SIP phone (registered with our
UC500 SIP proxy server and noted as 8778) calling number 7700 including another active
SIP call to number 19252302203.
SIP UAC CALL INFO


SIP UAS CALL INFO

Call 1
SIP Call ID : 461b79ec6e6089130a3dbacc2b196226@10.10.10.146
State of the call : SIP_STATE_OPTIONS_WAIT (27)

Calling Number : asterisk
Bit Flags : 0x40000C 0x104 0x0
CC Call ID : 63829
Media Stream 1
State of the stream : STREAM_IDLE
Stream Call ID : -1
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice

Call 2
SIP Call ID : 73d743e7-c8fa-1810-90d0-0019d111d2e8@mat-dtop01tra
Bit Flags : 0xC0401C 0x100 0x4
CC Call ID : 63823
Media Stream 1
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Media Stream 2
State of the stream : STREAM_DEAD
Stream Call ID : -1
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)



6.1.5 External Calling summary
In this example, we have placed an external call through our PSTN network, not SIP trunk,
and this section will show some of the detailed captures.

This command shows a summary of voice calls. Here we have an IP Phone connected to
virtual port 50/0/10 and our PSTN analog line is connected to FXO port 0/1/0. We can see
that a call has been connected using G711 CODEC.
uc01tra#show voice call summary
PORT CODEC VAD VTSP STATE VPM STATE
============== ========= === ==================== ======================
0/0/0 - - - FXSLS_ONHOOK
0/1/0 g711ulaw n S_CONNECT FXOLS_OFFHOOK
0/1/1 - - - FXOLS_ONHOOK
0/4/0 - - - EM_ONHOOK
0/4/1 - - - EM_ONHOOK
50/0/10 .1 g711ulaw n S_CONNECT EFXS_CONNECT
50/0/10 .2 - - - EFXS_ONHOOK
50/0/20 .1 - - - EFXS_ONHOOK
50/0/21 .1 - - - EFXS_ONHOOK
50/0/11 .1 - - - EFXS_ONHOOK
50/0/12 .1 - - - EFXS_ONHOOK
50/0/13 .1 - - - EFXS_ONHOOK
50/0/13 .2 - - - EFXS_ONHOOK
50/0/1 .1 - - - EFXS_ONHOOK
50/0/2 .1 - - - EFXS_ONHOOK
50/0/22 .1 - - - EFXS_ONHOOK
50/0/22 .2 - - - EFXS_ONHOOK
50/0/23 .1 - - - EFXS_ONHOOK
50/0/23 .2 - - - EFXS_ONHOOK
50/0/24 .1 - - - EFXS_ONHOOK
50/0/24 .2 - - - EFXS_ONHOOK
50/0/25 .1 - - - EFXS_ONHOOK
50/0/25 .2 - - - EFXS_ONHOOK

This command shows another view of a voice call very similar to the previous command.
The information is pretty much the same, but this shows the dialed number we called across
our PSTN network.
uc01tra#show voice call status
CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers
0xF959 35ED 0x863831C0 50/0/10.0 2091117170 g711ulaw 20001/100
0xF95A 35ED 0x8584E568 0/1/0 0/1:1 *2091117170 g711ulaw 100/20001
1 active call found

Looking at our IP Phones, we see that ephone profile 6 is off the hook with an active call
established.
uc01tra#show ephone summary

hairpin_block:
ephone-1 Mac:001B.D52C.77C5 TCP socket:[-1] activeLine:0 DECEASED
IP:10.67.78.128 7906 keepalive 114 1:13

ephone-5 Mac:0012.00A7.72EA TCP socket:[3] activeLine:0 REGISTERED
IP:10.67.78.36 Telecaster 7960 keepalive 1676 1:10 2:12


ephone-6 Mac:0011.932B.8B15 TCP socket:[2] activeLine:1 REGISTERED
IP:10.67.78.31 7970 keepalive 3718 1:10 2:11 3:13

ephone-7 Mac:000D.288E.3F4A TCP socket:[1] activeLine:0 REGISTERED
IP:10.67.78.111 7920 keepalive 53081 1:10 2:12

Max 14, Registered 3, Unregistered 0, Deceased 1, Sockets 4
ephone_send_packet process switched 0

Max Conferences 8 with 0 active (8 allowed)
Skinny Music On Hold Status
Active MOH clients 0 (max 156), Media Clients 0, B-ACD Clients 0
File music-on-hold.au type AU Media_Payload_G711Ulaw64k 160 bytes

6.1.6 Email Notification and Voice Messaging (CUE)
The other network configuration guide focused more with Voice messaging on our
UC500/CUE module. Here are focused on enabling email notifications where we can receive
an email with an attachment with the voicemail we received. Below are those show captures
along with some of the configuration that is not shown within the CUE configuration.

Below reflects the additional configuration needed for enabling email voicemail notification on
the CUE within our UC500 appliance.
username RouteHub profile RouteHub-vm-profile email address sales@RouteHub.com
username RouteHub profile RouteHub-vm-profile email enable
username RouteHub profile RouteHub-vm-profile email preference all
username RouteHub profile RouteHub-vm-profile email attach
username RouteHub profile RouteHub-vm-profile email schedule day 1 active from 01:00 to 24:00

This command shows a summary of whether voicemail notification is enabled and other
enabled details like attach voice message, which we configured.
cue01tra# show voicemail notification
Message Notification: enabled
Notification Preference: All
Connection Timeout: 48 seconds
Login to VoiceMail allowed: no
Attach voice message: yes

A voicemail profile needs to be created for a user and this shows the profile created, which is
RouteHub-vm-profile and if it is enabled.
cue01tra# show voicemail notification owner RouteHub
Profile: RouteHub-vm-profile

This command is very similar to the last, but is more focused on the type of notification
enabled. In our case we have email notification enabled. You can see the email address
voice messages will go to, if it is enabled, and if the voicemail message will be attached. You
can also see the other notification options available.
cue01tra# show voicemail notification owner RouteHub profile

Device Status Preference Num/Email ExtraDigits AttachVM
---------------------------------------------------------------------------------
Home phone disabled Urgent
Work phone disabled Urgent
Cell phone disabled All 812091117170
Numeric Pager disabled Urgent
Text Pager disabled Urgent
Email inbox enabled All sales@RouteHub.com YouHaveANewVoiceMailyes

This is another voicemail notification command for viewing the active schedule on when
emails with voicemail attachments are allowed to be sent. This is probably the most
important section and troubleshooting required for voice notification via email to work. When
you set this up and you leave a message no email may arrive. This is likely due to the active
schedule and time the system is allowed to send emails. Adjust the schedule accordingly
that best fits with what you are after.
cue01tra# show voicemail notification owner RouteHub email
Device: Email Inbox
Enabled: Yes
Preference: All
Email address: sales@RouteHub.com
Text: YouHaveANewVoiceMail
Attach VM: Yes
Schedule(active hours):
Sunday: 01:00 to 24:00
Monday: 01:00 to 24:00
Tuesday: 01:00 to 24:00
Wednesday: 01:00 to 24:00
Thursday: 01:00 to 24:00
Friday: 01:00 to 24:00
Saturday: 01:00 to 24:00

This command shows a summary of all mailboxes on Cisco Unity Express to include the
number of messages, mailbox usage, which messages are saved and more.
cue01tra# show voicemail mailboxes
OWNER MSGS NEW SAVE DEL BCST FUTR MSGTIME MBXSIZE USED
"dn6700" 1 0 1 0 0 0 17 420 4 %
"mthomati" 1 1 0 0 0 0 25 420 6 %
"RouteHub" 0 0 0 0 0 0 20 500 4 %
"ucm6_mthomati" 0 0 0 0 0 0 0 100 0 %
"ucm6_RouteHub" 0 0 0 0 0 0 0 100 0 %
"ucm6_sip1" 0 0 0 0 0 0 0 100 0 %
cue01tra# show voicemail users
"dn6700,/sw/local/users"
"mthomati,/sw/local/users"
"RouteHub,/sw/local/users"
"ucm6_mthomati,/sw/local/users"
"ucm6_RouteHub,/sw/local/users"
"ucm6_sip1,/sw/local/users"
cue01tra# show voicemail usage
personal mailboxes: 6
general delivery mailboxes: 0
orphaned mailboxes: 0
capacity of voicemail (minutes): 840
allocated capacity (minutes): 27.333333333333332
total message time used (seconds): 35
total message count: 3
average message length (seconds): 11.666666666666666
broadcast message count: 0
future message count: 0
networking message count: 0
greeting time used (seconds): 25
greeting count: 2
average greeting length (seconds): 12.5
total time used (seconds): 60

total time used (minutes): 1.0
percentage used time (%): 1
messages left since boot: 27
messages played since boot: 34
messages deleted since boot: 29

6.2 Troubleshooting

6.2.1 Root Causes

Once a network has been deployed and working operational any issue that will occur will
likely be due to one of the following below:
1. User Error
2. Software Error or Failure
3. Hardware Error or Failure
4. Power Error or Failure
5. Traffic Increase
6. Security Related
7. Third-Party Components

6.2.2 Initial questions to ask

Once a network has been deployed and working operational any issue that will occur will
likely be due to the following:
1. What has changed recently anywhere on the network?
a. Not just routers or switch, but with servers and various services such as
DNS, SMTP, etc. This tends to be the most common issue we have seen
where different groups make services changes like DNS, as an example, and
certain things on the network break where nothing was changed on the
routers or firewalls. However, the DNS changes affected some of the other
services on the network. That group who made the change will assume that
they didn't think that change would affect the network. Remember, IT is all
connected in more than one way, so validating all changes with all IT groups
is critical to confirm what could break including other considerations. Plus
any changes should rerun (or test) there baseline punch list to confirm that
all services outlined in the baseline are operational as they were before and
after any changes.
2. Confirm for any network changes? If so, check for configuration syntax errors and
cross check against a known working configuration.

6.2.3 Typical fixes

Identifying the root cause and resolving it are two separate things. Fixing a problem will
usually involve one or more of the following
Configuration change or rollback
Reboot
Software upgrade
Hardware replacement
It may require a configuration change or a rollback to a previously working configuration
known to work.
A reboot may do it or a software upgrade may be needed where a bug has emerged and/or a
hardware replacement may be needed, though is very rare.

7 Sample Full Configuration
7.1 CME and CUE on UC520

The following full configuration is for a Cisco UC520 appliance router. This is one of our
actual production routers including what we use for some of our demos. Many of the feature
configuration discussion in this workbook is shown actually including other core network
services such as routing and switching.

7.1.1 CME 7.1 on UC520

Current configuration : 47258 bytes
!
! Last configuration change at 17:57:32 PST Wed Dec 23 2009 by mthomati
! NVRAM config last updated at 15:04:01 PST Thu Dec 17 2009 by mthomati
!
version 12.4
parser config cache interface
service nagle
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service internal
service compress-config
service sequence-numbers
!
hostname uc01tra
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
logging buffered 16384
enable secret 5 $1$.OE9$peixHx./t0zcoXNKpw68z0
!
!
aaa authentication login default local
aaa authentication login console line
aaa authorization exec default local
!
!
aaa session-id common
!
monitor session 1 source interface Fa0/1/7
monitor session 1 destination interface Fa0/1/8
clock timezone PST -8
clock summer-time PDT recurring
!
crypto pki trustpoint TP-self-signed-781422512
enrollment selfsigned

subject-name cn=IOS-Self-Signed-Certificate-781422512
revocation-check none
rsakeypair TP-self-signed-781422512
!
crypto pki trustpoint Equifax_Secure_CA
!
crypto pki trustpoint NetworkSolutions_CA
!
crypto pki trustpoint trps1_server
!
!
crypto pki certificate chain TP-self-signed-781422512
certificate self-signed 01
3082024A 308201B3 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
30312E30 2C060355 04031325 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 37383134 32323531 32301E17 0D303930 37313030 39333334
315A170D 32303031 30313030 30303030 5A303031 2E302C06 03550403 1325494F
532D5365 6C662D53 69676E65 642D4365 72746966 69636174 652D3738 31343232
35313230 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D207FB3A 5F28207E 0A83D6F6 CEACF323 AD0AC7B2 920D1815 A304DAC4 9372BA55
9FA0A4EB E0EAA50A 21D70A8F 855C3BAB 7AAB90EC 80A13600 4310B8C0 9377678D
B1E222C6 F06C28B8 5BA19DD3 ACCCBA10 D053D828 B07EBF92 ABDF86B7 2B119217
51AA8D7A 890C4CD3 CA223CEE 48A002EF 81C31A14 3DB01926 D1F4A7CA F31763DB
02030100 01A37430 72300F06 03551D13 0101FF04 05300301 01FF301F 0603551D
11041830 16821475 63303174 72612E72 6F757465 6875622E 636F6D30 1F060355
1D230418 30168014 3D86C4EF A3175A13 313A16B9 A482C5B5 A19A1E74 301D0603
551D0E04 1604143D 86C4EFA3 175A1331 3A16B9A4 82C5B5A1 9A1E7430 0D06092A
864886F7 0D010104 05000381 810002A9 EE2F5747 3784FB27 9A34D23D 099369BB
F1F811EB F6609B96 EC536B99 1E5C8CAD B02B4B3C FCCAD113 D9B41DDB 1526317A
F4B066FB 6FB2D7A6 8CC29B6A 8D9C667D 8F56C49A 3A906851 C3B6E636 7E5E5FDF
903CD8B7 5B5B21D6 C54E6F37 F6D922E3 EA005478 8154A24A 82393601 164C91FF
A2E1D99A 7B7EFDB4 C090BEBD 8333
quit
crypto pki certificate chain Equifax_Secure_CA
certificate ca 35DEF4CF
30820320 30820289 A0030201 02020435 DEF4CF30 0D06092A 864886F7 0D010105
0500304E 310B3009 06035504 06130255 53311030 0E060355 040A1307 45717569
66617831 2D302B06 0355040B 13244571 75696661 78205365 63757265 20436572
74696669 63617465 20417574 686F7269 7479301E 170D3938 30383232 31363431
35315A17 0D313830 38323231 36343135 315A304E 310B3009 06035504 06130255
53311030 0E060355 040A1307 45717569 66617831 2D302B06 0355040B 13244571
75696661 78205365 63757265 20436572 74696669 63617465 20417574 686F7269
74793081 9F300D06 092A8648 86F70D01 01010500 03818D00 30818902 818100C1
5DB15867 0862EEA0 9A2D1F08 6D911468 980A1EFE DA046F13 846221C3 D17CCE9F
05E0B801 F04E34EC E28A9504 64ACF16B 535F05B3 CB6780BF 42028EFE DD0109EC
E100144F FCFBF00C DD43BA5B 2BE11F80 70991557 9316F10F 976AB7C2 68231CCC
4D5930AC 511E3BAF 2BD6EE63 457BC5D9 5F50D2E3 500F3A88 E7BF14FD E0C7B902
03010001 A3820109 30820105 30700603 551D1F04 69306730 65A063A0 61A45F30
5D310B30 09060355 04061302 55533110 300E0603 55040A13 07457175 69666178
312D302B 06035504 0B132445 71756966 61782053 65637572 65204365 72746966
69636174 65204175 74686F72 69747931 0D300B06 03550403 13044352 4C31301A
0603551D 10041330 11810F32 30313830 38323231 36343135 315A300B 0603551D
0F040403 02010630 1F060355 1D230418 30168014 48E668F9 2BD2B295 D747D823
20104F33 98909FD4 301D0603 551D0E04 16041448 E668F92B D2B295D7 47D82320
104F3398 909FD430 0C060355 1D130405 30030101 FF301A06 092A8648 86F67D07
4100040D 300B1B05 56332E30 63030206 C0300D06 092A8648 86F70D01 01050500
03818100 58CE29EA FCF7DEB5 CE02B917 B585D1B9 E3E095CC 25310D00 A6926E7F
B692639E 5095D19A 6FE411DE 63856E98 EEA8FF5A C8D355B2 667157DE C021EB3D
2AA72349 01048642 7BFCEE7F A21652B5 6767D340 DB3B2658 B228773D AE147761
D6FA2A66 27A00DFA A7735CEA 70F19421 65445FFA FCEF2968 A9A28779 EF79EF4F
AC077738

quit
crypto pki certificate chain NetworkSolutions_CA
certificate ca 10E776E8A65A6E377E050306D43C25EA
308204A6 3082038E A0030201 02021010 E776E8A6 5A6E377E 050306D4 3C25EA30
0D06092A 864886F7 0D010105 05003081 97310B30 09060355 04061302 5553310B
30090603 55040813 02555431 17301506 03550407 130E5361 6C74204C 616B6520
43697479 311E301C 06035504 0A131554 68652055 53455254 52555354 204E6574
776F726B 3121301F 06035504 0B131868 7474703A 2F2F7777 772E7573 65727472
7573742E 636F6D31 1F301D06 03550403 13165554 4E2D5553 45524669 7273742D
48617264 77617265 301E170D 30363034 31303030 30303030 5A170D32 30303533
30313034 3833385A 3062310B 30090603 55040613 02555331 21301F06 0355040A
13184E65 74776F72 6B20536F 6C757469 6F6E7320 4C2E4C2E 432E3130 302E0603
55040313 274E6574 776F726B 20536F6C 7574696F 6E732043 65727469 66696361
74652041 7574686F 72697479 30820122 300D0609 2A864886 F70D0101 01050003
82010F00 3082010A 02820101 00C3DD36 CC83C318 55B096D9 1325D326 864838BB
167FF19F 29F6FD03 F1ED4D26 9A56F0B5 1A1ACDE6 CC855540 A4B5D00D CA22EF3D
23C67E6C CCBCA1E9 7C5046E0 BD14AD65 12C20B11 69520A07 921F736F C1BAD762
F0CE002E 34A5C8E6 2F0FEC0D EA446175 68E5E4DC 80364FDA 785D5325 9494F54F
2E3A606F 0CA6D9B3 F62A2E03 12D52642 0751B264 5771DC21 1C89C769 A3E6FBC2
7B6EEF0C 87FB5064 E84E4BEF E7719B83 6361C932 8D8CEC14 A7E489AD 3F2B2664
E48542F2 8950E13A BE15E345 25E25ACB 8C3FE033 1E35095A 84EA7E5D A1F59180
0A2806B7 CB314125 618B01E9 56A2F63E 5F2FF3C4 43F61994 75834CA1 82423AC6
BAC40930 A6E17502 51B95E64 8B020301 0001A382 01203082 011C301F 0603551D
23041830 168014A1 725F261B 28984395 5D0737D5 85969D4B D2C34530 1D060355
1D0E0416 04143C41 E28F0808 A94C2589 8D6DC538 D0FC858C 6217300E 0603551D
0F0101FF 04040302 01063012 0603551D 130101FF 04083006 0101FF02 01003019
0603551D 20041230 10300E06 0C2B0601 0401860E 01020103 01304406 03551D1F
043D303B 3039A037 A0358633 68747470 3A2F2F63 726C2E75 73657274 72757374
2E636F6D 2F55544E 2D555345 52466972 73742D48 61726477 6172652E 63726C30
5506082B 06010505 07010104 49304730 4506082B 06010505 07300286 39687474
703A2F2F 7777772E 75736572 74727573 742E636F 6D2F6361 63657274 732F5554
4E416464 54727573 74536572 7665725F 43412E63 7274300D 06092A86 4886F70D
01010505 00038201 010068AB FCEF806B 18B2B0B3 A34589CB 53C5A2E6 AF08A9FD
FF0F49AC FFE49FD7 417CA3C5 A2E8AAE0 57212DC3 AA7C0C4C 280B79F4 EE4C32AD
790E7EA2 5E34184F DF54F1BD 687CE3D3 D7465E6D 64C2F76D 8882730C EF9985EA
A9EF324A F0839F73 910CA43E 2B3151A6 628F1584 F9A63A12 303FDA6E F8CCC719
920F5CF4 FE17F195 0847522C 508FE89B A5EEAE70 33899182 FE30AA76 7659D76C
18D32B12 5B1D281D 7871F6CD 36A2E907 48443BE7 576E820A ADC58ADD E853B471
AF13D206 9D376D53 3F8A3508 FAFEA216 E6B96F5C 5639D6C6 AAEF1967 CE13C5B8
9505FB0A 44C99FA9 40254B32 11AF07FE 08D54271 E9E1538B 151FDD2A 07957024
6F645ED3 B7902E8B 21D8
quit
crypto pki certificate chain trps1_server
certificate ca 00
3082029F 30820208 02010030 0D06092A 864886F7 0D010104 05003081 97310B30
09060355 04061302 55533111 300F0603 55040813 08436F6C 6F726164 6F311030
0E060355 04071307 426F756C 64657231 16301406 0355040A 130D4369 73636F20
53797374 656D7331 0C300A06 0355040B 13035354 47311D30 1B060355 04031314
74727073 312D626C 64722E63 6973636F 2E636F6D 311E301C 06092A86 4886F70D
01090116 0F777473 75694063 6973636F 2E636F6D 301E170D 30363130 32333230
32363231 5A170D30 39303731 39323032 3632315A 30819731 0B300906 03550406
13025553 3111300F 06035504 08130843 6F6C6F72 61646F31 10300E06 03550407
1307426F 756C6465 72311630 14060355 040A130D 43697363 6F205379 7374656D
73310C30 0A060355 040B1303 53544731 1D301B06 03550403 13147472 7073312D
626C6472 2E636973 636F2E63 6F6D311E 301C0609 2A864886 F70D0109 01160F77
74737569 40636973 636F2E63 6F6D3081 9F300D06 092A8648 86F70D01 01010500
03818D00 30818902 818100BF F80B7E13 19C5AA37 D7433EDC 4EC5CAD8 40BEE950
7C099395 997043C9 B9C4BCF6 DF97F091 0ECB7D06 F1B336C6 CD134A67 826B0182
09535A4B 11EB4BE8 B46187CB BBD9FECB CB03AE65 8F2C5E7E 40A66FF2 899E2FF1
CBC072B2 A9B537C0 84C9F873 8A141ED9 D8D15186 F7047400 BB8A2CA1 C59DEAD8
DA09FBB3 6E67D8BF F6811102 03010001 300D0609 2A864886 F70D0101 04050003
818100AC C6185869 1324F6BD 728A8D00 CEDF15E3 14671016 90ED8F7B 5FF72860
8F9469D2 B344641D 75E4A566 BCB06ACE 21DFC2B3 041A961C 8A23610A 284BC399
8E632BBA C734D76A 266E6A45 88DC366F C5E12E9E 087AC3AA 7FEE2089 C97821A7

882BFEC3 26425299 11700277 B9E4EBCD 15A0B388 F8D4A102 E472A398 63E0D7DA
5BFBE1
quit
dot11 syslog
!
dot11 ssid wpa-psk
authentication open
authentication key-management wpa
wpa-psk ascii 7 -removed--
!
ip cef
!
!
ip dhcp relay information trust-all
ip dhcp excluded-address 10.67.78.1 10.67.78.29
!
ip dhcp pool RHG-DHCP-LAN-POOL
network 10.67.78.0 255.255.255.0
option 150 ip 10.67.78.1
dns-server 206.13.28.12 64.169.140.6
lease infinite
!
!
no ip bootp server
ip domain lookup source-interface BVI10
ip domain name routehub.com
ip name-server 206.13.28.12
ip multicast-routing
ip reflexive-list timeout 120
!
!
!
stcapp ccm-group 1
stcapp
!
stcapp feature access-code
!
!
!
!
multilink bundle-name authenticated
!
!
voice call send-alert
voice call carrier capacity active
!
voice service voip
sip
registrar server expires max 3600 min 3600
localhost dns:sanfrancisco-1.vtnoc.net
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!

!
!
!
!
!
!
voice class custom-cptone routehub-join
dualtone conference
frequency 1200 1200
cadence 150 50 150 50
!
voice class custom-cptone routehub-leave
dualtone conference
frequency 900 900
cadence 150 50 150 50
!
!
!
!
!
!
mode cme
max-dn 12
max-pool 12
timezone 47
time-format 24
date-format YY-M-D
!
voice register dn 1
number 8701
name Routehub SIP client (X-lite)
!
voice register dn 2
number 8778
name Michel Thomatis (SIP)
!
id mac 000C.F179.1682
number 1 dn 1
codec g711ulaw
!
id mac 0019.D111.D2E8
number 1 dn 2
codec g711ulaw
!
pin 6778
number 6700,A5001,7700,1001,2001 type feature-ring
!
pin 6700
!
pin 6778

user 78 password 78
number 6700,A5001,7700,1001,2001 type feature-ring
!
pin 6700
user 70 password 70
!
!
rule 1 /^8(.*)/ /\1/
rule 2 /^8$1[2-9].........$$/ /\1/
!
rule 1 /^.*/ /19252302203/
!
rule 1 /^4$91[2-9].........$$/ /\1/
!
rule 1 reject /8002197425/
rule 2 reject /2402107113/
rule 3 reject /4345339030/
!
rule 1 /^.*$......$/ /\1/
!
rule 1 /19252302203/ /7700/
!
rule 1 /^.*$....$/ /\1/
!
rule 1 /^.*$....$/ /\1/
!
!
voice translation-profile call_block
translate calling 5
!
voice translation-profile routehub-tp-ucm7-outgoing
translate called 192
!
voice translation-profile routehub-tp-s3-external
translate called 4
!
translate calling 3
translate called 2
!
voice translation-profile routehub-tp-sip-outgoing2
translate called 13
!
voice translation-profile routehub-tp-ucm6
translate called 10
!
voice translation-profile routehub-tp-ucm7-outgoing
translate called 191
!
!
voice-card 0
dspfarm
dsp services dspfarm
!

fax interface-type fax-mail
mta send server 10.67.78.6 port 25
mta send subject You Received a Fax!
mta send with-subject both
mta send postmaster sales@routehub.com
mta send mail-from hostname routehub.com
mta send mail-from username IncomingFax
mta send return-receipt-to hostname routehub.com
mta send return-receipt-to username michel
mta receive aliases routehub.com
mta receive aliases 10.67.78.6
mta receive maximum-recipients 10
mta receive generate permanent-error
!
application
service aa flash:app-b-acd-aa-2.1.2.3.tcl
param aa-hunt2 6721
paramspace english index 1
param queue-len 5
param handoff-string aa
param dial-by-extension-option 1
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 6720
paramspace english location flash:
param second-greeting-time 30
param call-retry-timer 15
param voice-mail 6000
param max-time-call-retry 300
param service-name queue
!
service onramp flash:app_faxmail_onramp.2.0.1.3.tcl
!
service fax_detect flash:app_fax_detect.2.1.2.2.tcl
param fax-dtmf 2
param mode listen-first
param voice-dtmf 1
!
service queue flash:app-b-acd-2.1.2.3.tcl
param queue-len 5
param aa-hunt2 6721
!
!
!
!
!
spanning-tree backbonefast
spanning-tree vlan 1 priority 1000
vtp domain ROUTEHUB
vtp mode transparent
username mthomati privilege 15 secret 5 $1$.mR8$tKhJRqb8V0JpofbXKIlpg0
!
!
!

archive
log config
hidekeys
!
!
vlan 10
name RHG-VLAN
!
vlan 11
name RHG-VLAN11
!
vlan 999
name RHG-VLAN-BITBUCKET
shutdown
!
ip tcp synwait-time 10
ip telnet source-interface Vlan10
ip telnet quiet
ip telnet hidden addresses
ip tftp source-interface Vlan10
ip ssh source-interface Vlan10
ip ssh version 2
!
class-map match-any AutoQoS-VoIP-RTP-Trust
match ip dscp ef
class-map match-any AutoQoS-VoIP-Control-Trust
match ip dscp cs3
match ip dscp af31
!
!
policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority percent 70
class AutoQoS-VoIP-Control-Trust
bandwidth percent 5
class class-default
fair-queue
!
bridge irb
!
!
!
interface Null0
no ip unreachables
!
description $FW_OUTSIDE$
ip address 169.1.1.73 255.255.255.248
ip access-group ingress-acl in
ip access-group egress-acl out
ip verify unicast reverse-path
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
ip flow egress
ip nat outside
load-interval 30
speed 100
full-duplex
snmp trap ip verify drop-rate
no cdp enable
crypto map ezvpn

!
description RHG: CUE interface
ip unnumbered BVI10
ip nat inside
service-module ip address 10.67.78.2 255.255.255.0
service-module ip default-gateway 10.67.78.1
!
macro description cisco-phone
auto qos voip trust
service-policy output AutoQoS-Policy-Trust
!
load-interval 30
macro description cisco-desktop
!
load-interval 30
duplex full
auto qos voip trust
!
load-interval 30
!
load-interval 30
!
load-interval 30
!
load-interval 30
auto qos voip trust
!
switchport trunk native vlan 10
switchport trunk allowed vlan 1,2,10,11,1002-1005

!
load-interval 30
macro description cisco-switch
!
interface Dot11Radio0/5/0
no ip address
!
encryption mode ciphers tkip
!
ssid wpa-psk
!
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
channel 2437
station-role root
rts threshold 2312
no cdp enable
bridge-group 10
bridge-group 10 subscriber-loop-control
bridge-group 10 block-unknown-source
no bridge-group 10 source-learning
no bridge-group 10 unicast-flooding
!
interface Virtual-Template1
no ip address
no keepalive
!
interface Vlan1
no ip address
bridge-group 1
!
interface Vlan10
no ip address
no ip redirects
no ip unreachables
no ip proxy-arp
bridge-group 10
!
interface Vlan11
no ip address
no ip redirects
bridge-group 11
!
interface Vlan100
no ip address
bridge-group 100
!
interface BVI1
no ip address
!
interface BVI10
ip address 192.168.0.1 255.255.255.0 secondary
ip address 10.67.78.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
ip flow egress
ip pim sparse-mode
ip nat inside

!
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 169.1.1.78
ip route 10.67.78.2 255.255.255.255 Integrated-Service-Engine0/0
ip flow-cache timeout active 1
ip flow-export source BVI10
ip flow-export version 5
ip flow-export destination 10.67.78.52 9996
ip flow-top-talkers
top 5
sort-by bytes
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
ip dns server
ip pim rp-address 10.67.78.1 Class-D-Space override
ip nat inside source list RHG-acl-nonat interface FastEthernet0/0 overload
!
ip access-list standard Class-D-Space
permit 224.0.0.0 15.255.255.255
ip access-list standard RHG-acl-netmgmt
permit 10.67.78.0 0.0.0.255
deny any log
!
ip access-list extended egress-acl
permit icmp any any reflect reflexive-acl
permit tcp any any reflect reflexive-acl
permit udp any any reflect reflexive-acl
permit esp host 169.1.1.73 any
permit gre host 169.1.1.73 any
permit gre any any
deny ip any any log
ip access-list extended ingress-acl
deny ip 10.0.0.0 0.255.255.255 any
deny ip 172.16.0.0 0.15.255.255 any
deny ip 192.168.0.0 0.0.255.255 any
permit icmp any any echo-reply
permit icmp any any time-exceeded
permit udp any eq domain any
permit tcp any host 169.1.1.77 eq 3389 8080 www smtp 81 22
permit tcp any host 169.1.1.73 eq 1723 telnet 22 4662 10000
permit tcp any host 169.1.1.77 eq 5800 5900
permit tcp any host 169.1.1.77 eq 9996
permit udp any host 169.1.1.73 eq non500-isakmp
permit tcp any eq ftp-data host 169.1.1.73
permit tcp any host 169.1.1.75 eq ftp
permit udp any eq ntp any
permit udp any host 169.1.1.73 eq isakmp
permit esp any host 169.1.1.73
permit gre any any
permit tcp host 74.93.5.82 eq ftp-data any
evaluate reflexive-acl
deny ip any any log
ip access-list extended RHG-acl-nonat
permit ip 10.67.78.0 0.0.0.255 any
permit ip 192.168.5.0 0.0.0.255 any
permit ip 192.168.10.0 0.0.0.255 any
permit ip 192.168.0.0 0.0.0.255 any
permit ip 10.67.79.0 0.0.0.255 any
!

logging trap debugging
logging 10.67.78.8
access-list 10 permit 192.168.1.0 0.0.0.255
snmp-server community RHG-snmp RO RHG-acl-netmgmt
snmp-server ifindex persist
snmp-server contact RouteHub Group
mac-address-table aging-time 896
!
!
!
!
tftp-server flash:CP7902080001SCCP051117A.sbin
tftp-server flash:apps11.1-1-2-26.sbn
tftp-server flash:cnu11.3-1-2-26.sbn
tftp-server flash:cvm11sccp.8-0-2-25.sbn
tftp-server flash:dsp11.1-1-2-26.sbn
tftp-server flash:jar11sccp.8-0-2-25.sbn
tftp-server flash:SCCP11.8-0-3S.loads
tftp-server flash:term06.default.loads
tftp-server flash:S00104000100.sbn
tftp-server flash:cmterm_7936.3-3-5-0.bin
tftp-server flash:P0030702T023.loads
tftp-server flash:P0030702T023.sbn
tftp-server flash:P0030702T023.sb2
tftp-server flash:P0030702T023.bin
tftp-server flash:cmterm-7941-7961-sccp.7.0.3.tar
tftp-server flash:CVM41.2-0-2-26.sbn
tftp-server flash:Jar41.2-9-2-26.sbn
tftp-server flash:TERM41.7-0-3-0S.loads
tftp-server flash:CVM70.2-0-2-26.sbn
tftp-server flash:Jar70.2-9-2-26.sbn
tftp-server flash:TERM70.7-0-3-0S.loads
tftp-server flash:SCCP31.8-1-1SR2S.loads
tftp-server flash:APPS-1.0.1.SBN
tftp-server flash:CP7921G-1.0.1.LOADS
tftp-server flash:GUI-1.0.1.SBN
tftp-server flash:SYS-1.0.1.SBN
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:SCCP70.8-3-4SR1S.loads
tftp-server flash:RingList.xml
tftp-server flash:DistinctiveRingList.xml
tftp-server flash:24.raw
tftp-server flash:24ad.raw

tftp-server flash:Desktops/320x212x12/routehub-full.png
tftp-server flash:Desktops/320x212x12/routehub-tmb.png
!
control-plane
!
bridge 1 protocol ieee
bridge 1 route ip
bridge 10 route ip
bridge 11 route ip
bridge 100 route ip
bridge 101 route ip
bridge 102 route ip
bridge 255 route ip
!
!
voice-port 0/0/0
timeouts ringing infinity
station-id name from Analog
caller-id enable
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
voice-port 0/1/0
pre-dial-delay 0
no vad
caller-id enable
!
voice-port 0/1/1
caller-id enable
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
!
sccp local BVI10
sccp
!
sccp ccm group 1
keepalive retries 5

!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
conference-join custom-cptone routehub-join
conference-leave custom-cptone routehub-leave
associate application SCCP
!
dial-peer cor custom
name internal
name local
name domestic
name international
name 900
name 976
name 1209627
name 209627
!
!
!
call-block translation-profile incoming call_block
call-block disconnect-cause incoming call-reject
direct-inward-dial
port 0/1/0
!
codec g711ulaw
no vad
!
destination-pattern 8[2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
destination-pattern 8[0-1][2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
!

description **Emergency Outgoing Call to SIP Trunk**
voice-class codec 1
!
translation-profile outgoing routehub-tp-ucm7-outgoing
dtmf-relay sip-kpml
codec g711ulaw
!
service stcapp
port 0/0/0
!
description RP-9011! INTERNATIONAL
destination-pattern 9.[0][1][1]T
port 0/1/0
!
description ** UNIFIED MESSAGING **
destination-pattern 671.
dtmf-relay rtp-nte
codec g711ulaw
!
incoming called-number 800[0,1].....
codec g711ulaw
no vad
!
service aa
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
description FAX
!
service onramp
direct-inward-dial
port 0/1/1
!
translation-profile outgoing routehub-tp-ucm7-outgoing

dtmf-relay sip-kpml
codec g711ulaw
!
!
sip-ua
authentication username 19252302203 password 7 -removed--
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sanfrancisco-1.vtnoc.net expires 3600
sip-server dns:sanfrancisco-1.vtnoc.net
host-registrar
!
!
!
telephony-service
sdspfarm conference mute-on 11 mute-off 12
sdspfarm units 3
sdspfarm tag 1 mtp001d4567c690
conference hardware
video
em logout 0:0 0:0 0:0
max-ephones 14
max-dn 56
calling-number initiator
timeouts interdigit 5
url services http://10.67.78.2/voiceview/common/login.do
url authentication
http://10.67.78.2/voiceview/authentication/authenticate.do
load 7914 S00104000100
load 7902 CP7902080001SCCP051117A
load 7906 SCCP11.8-0-3S
load 7911 SCCP11.8-0-3S
load 7921 CP7921G-1.0.1
load 7936 cmterm_7936.3-3-5-0
load 7960-7940 P0030702T023
load 7941 TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S
load 7971 TERM70.7-0-3-0S
time-zone 5
live-record 6005
voicemail 6000
web admin system name admin secret 5 $1$ucO7$XfKgADX7L1nzz11jbTDa./
dn-webedit
time-webedit

directory entry 1 919252302203 name ROUTEHUB GROUP (Main)
directory entry 2 912098329950 name Deliver Ease (Main)
fac standard
!
!
ephone-template 1
softkeys hold Newcall Resume Select Join
softkeys idle Redial Newcall Dnd Cfwdall Pickup ConfList
softkeys seized Redial Endcall Pickup Meetme
softkeys connected Endcall LiveRcd Confrn Hold Park Trnsfer TrnsfVM
!
!
ephone-dn 1
number 6001
name RouteHub Paging System
paging ip 239.192.2.1 port 2000
!
!
ephone-dn 2
number 6002
park-slot timeout 30 limit 10
name RouteHub Call Park
!
!
ring internal
label 1001 (Office)
name 1001
!
!
ring internal
label 1002 (Family Room)
name 1002
mobility
!
!
ephone-dn 5
number 3001
!
!
ring external
label 6700 (Main)
description 2098322559
name 6700
!
!
ephone-dn 11
label Intercom
name Intercom
intercom A5002

!
!
ephone-dn 12
label Intercom
name Intercom
intercom A5001
!
!
ring external
number 7700 secondary 19252302203
label 7700 (RouteHub Main)
name 7700
!
!
number 2001
label 2001 (Agent)
!
!
number 2002
label 2002 (Agent)
!
!
ephone-dn 16
number 6005
!
!
ephone-dn 20
mwi on
!
!
ephone-dn 21
mwi off
!
!
number 6999
conference meetme
no huntstop
!
!
number 6999
conference meetme
preference 1
no huntstop
!
!
number 6999
conference meetme
preference 2
no huntstop
!
!

number 6999
conference meetme
preference 3
no huntstop
!
!
number 6998
name Conference
conference ad-hoc
preference 1
no huntstop
!
!
number 6998
name Conference
conference ad-hoc
preference 2
no huntstop
!
!
ephone 1
mac-address 001B.D52C.77C5
type 7906
button 1:13
!
!
!
ephone 2
video
mac-address 001C.58F0.7619
paging-dn 1
type 7970
logout-profile 2
!
!
!
ephone 3
mac-address 000C.299C.8EE7
type CIPC
!
!
!
ephone 4
mac-address D456.7C69.0000
max-calls-per-button 2
type anl
button 1:10
!
!
!
ephone 5
mac-address 0012.00A7.72EA
username "dn6700"
paging-dn 1
type 7960
button 1:10 2:12
!

!
!
ephone 6
video
mac-address 0011.932B.8B15
ephone-template 1
fastdial 1 1002 name FR1002
type 7970
no dnd feature-ring
logout-profile 1
!
!
!
ephone 7
mac-address 001A.2FE7.9B96
type 7960
button 1:14
!
!
ephone-hunt 1 longest-idle
pilot 6721
list 2001, 2002
timeout 10, 10
statistics collect
!
!
banner motd ^C
-----------------------------------------------------------------------
Powered by...
|| ||
|| ||
|||| ||||
..:||||||:..:||||||:..
OFFICAL USE ONLY!
RouteHub Group, LLC
(925) 230-2203
www.routehub.com
-----------------------------------------------------------------------
^C
alias exec c config t
alias exec acl show access-list
alias exec tel show run | b telephony-service
!
line con 0
exec-timeout 15 0
password 7 13543A42051F107939
logging synchronous
login authentication console
no modem enable
length 50
notify
transport preferred none
transport output all
stopbits 1
line aux 0
line 2
no activation-character
no exec
transport input all

line vty 0 4
transport input telnet ssh
transport output all
line vty 5 100
!
scheduler allocate 4000 400
scheduler interval 500
ntp source BVI10
ntp server 69.31.13.210 prefer
!
webvpn gateway gateway_1
ip address 169.1.1.73 port 443
ssl trustpoint TP-self-signed-781422512
inservice
!
webvpn install svc flash:/webvpn/svc_1.pkg sequence 1
!
webvpn context webvpn
title "ROUTEHUB SSL VPN"
secondary-color white
title-color #CCCC66
text-color black
ssl authenticate verify all
!
login-message "RouteHub Group Use Only"
!
policy group policy_1
functions svc-enabled
banner "RouteHub Group Use Only!"
svc address-pool "RHG-ezvpn-pool"
svc default-domain "routehub.com"
svc keep-client-installed
svc split include 10.67.78.0 255.255.255.0
svc split include 192.168.0.0 255.255.255.0
svc dns-server primary 10.67.78.6
default-group-policy policy_1
aaa authentication list sslvpn-aaa
gateway gateway_1
inservice

7.1.2 CUE 7.0.1 on UC520

clock timezone America/Los_Angeles

hostname cue01tra

ip domain-name routehub.com

line console
exit

system language preferred "en_US"


ntp server 69.31.13.210 prefer

software download server url "ftp://10.67.78.243/cue7" credentials hidden
"c9yem6kD1vLmzOwzb80QtWTRRNhfheJsSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9

J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP"
site name local
site-hostname 10.67.78.1
web credentials hidden
"IRBD7mOWpxpWaKsZ0zTwvt+jA8JZJIp2Sd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9
end site

fax gateway outbound address 10.67.78.1

privilege ViewPrivateList create
privilege ViewRealTimeReports create
privilege manage-passwords create
privilege ManagePrompts create
privilege broadcast create
privilege vm-imap create
privilege ManagePublicList create
privilege manage-users create
privilege ViewHistoricalReports create
privilege local-broadcast create

groupname Broadcasters create

username ucm6_routehub create
username ucm6_sip1 create
username administrator create
username admin create
username cisco create
username routehub create
username Fax create
username ucm6_mthomati create
username mthomati create

privilege ViewPrivateList description "Privilege to view private list"
privilege ViewRealTimeReports description "Privilege to view realtime
reports"
privilege manage-passwords description "Privilege to reset user passwords"
privilege ManagePrompts description "Privilege to create, modify, or delete
system prompts"
privilege broadcast description "Privilege to send local or remote broadcast
messages"
privilege vm-imap description "Privilege to manage personal voicemail via
IMAP client"
privilege ManagePublicList description "Privilege to manage public lists"
privilege manage-users description "Privilege to create, modify, and delete
users and groups"
privilege ViewHistoricalReports description "Privilege to view historical
reports"
privilege local-broadcast description "Privilege to send local broadcast
messages"
privilege ViewPrivateList operation voicemail.lists.private.view
privilege ViewRealTimeReports operation report.realtime
privilege manage-passwords operation user.pin
privilege manage-passwords operation user.password
privilege manage-passwords operation system.debug
privilege ManagePrompts operation system.debug
privilege ManagePrompts operation prompt.modify
privilege broadcast operation system.debug
privilege broadcast operation broadcast.remote
privilege broadcast operation broadcast.local
privilege vm-imap operation voicemail.imap.user
privilege ManagePublicList operation voicemail.lists.public
privilege ManagePublicList operation system.debug

privilege manage-users operation user.pin
privilege manage-users operation group.configuration
privilege manage-users operation user.mailbox
privilege manage-users operation user.configuration
privilege manage-users operation user.notification
privilege manage-users operation user.password
privilege manage-users operation system.debug
privilege manage-users operation user.remote
privilege ViewHistoricalReports operation report.historical.view
privilege local-broadcast operation system.debug
privilege local-broadcast operation broadcast.local

groupname Administrators member admin
groupname Administrators member routehub
groupname Administrators member administrator
groupname Administrators member mthomati
groupname Administrators member cisco
groupname Administrators member dn6700
groupname Broadcasters privilege broadcast

username ucm6_routehub phonenumber "106701"
username ucm6_sip1 phonenumber "106801"
username dn6700 phonenumber "6700"
username routehub phonenumber "7700"
username Fax phonenumber "6730"
username ucm6_mthomati phonenumber "106778"
username mthomati phonenumber "6778"
username routehub phonenumberE164 "19252302203"

restriction msg-notification create
restriction msg-notification min-digits 1
restriction msg-notification max-digits 30
restriction msg-notification dial-string preference 1 pattern * allowed

smtp server address ht1.routehub.com authentication credentials
b8Zr/zf3KMbC4a9xaoW7MKqBQzxbLoUpStTTbL/6Z6ZiprrDCoklfUnfGWTYHfmPSd8ZZNgd+Y9J
3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP

backup server url "ftp://10.67.78.243/cue" credentials hidden
"c9yem6kD1vLmzOwzb80QtWTRRNhfheJsSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9

calendar biz-schedule systemschedule
open day 1 from 00:00 to 24:00
end schedule

ccn application autoattendant aa
description "autoattendant"
enabled
maxsessions 6
script "aasimple.aef"
parameter "BusinessClosedPrompt" "AASPlayExtensions.wav"
parameter "BusinessOpenPrompt" "AASPlayExtensions.wav"
parameter "allowExternalTransfers" "false"
parameter "MaxRetry" "3"
parameter "BusinessSchedule" "systemschedule"
parameter "HolidayPrompt" "AAHolidayPrompt.wav"
parameter "WelcomePrompt" "AAWelcome.wav"

parameter "PlayExtensionsPrompt" "AASPlayExtensions.wav"
parameter "ExtensionLength" "4"
end application

ccn application ciscomwiapplication aa
enabled
maxsessions 4
script "setmwi.aef"
end application

ccn application msgnotification aa
description "msgnotification"
enabled
maxsessions 6
script "msgnotify.aef"
parameter "DelayBeforeSendDTMF" "1"
end application

ccn application promptmgmt aa
description "promptmgmt"
enabled
maxsessions 1
script "promptmgmt.aef"
end application

ccn application voicemail aa
description "voicemail"
enabled
maxsessions 6
script "voicebrowser.aef"
parameter "uri" "http://localhost/voicemail/vxmlscripts/login.vxml"
end application

ccn engine
end engine

ccn reporting historical
database local
description "cue01tra"
end reporting

ccn subsystem sip
end subsystem

ccn trigger http urlname msgnotifytrg
application "msgnotification"
enabled
maxsessions 2
end trigger

ccn trigger http urlname mwiapp
application "ciscomwiapplication"
enabled
maxsessions 1
end trigger


enabled
maxsessions 6
end trigger

enabled
locale "en_US"
maxsessions 6
end trigger

enabled
idletimeout 5000
locale "en_US"
maxsessions 1
end trigger

service phone-authentication
end phone-authentication

service voiceview
enable
end voiceview

voicemail notification enable
voicemail notification preference all
voicemail notification allow-login
voicemail notification email attach

voicemail callerid
voicemail configuration outgoing-email from-address sales@routehub.com
voicemail default language en_US
voicemail default mailboxsize 420
voicemail broadcast recording time 300
voicemail default messagesize 240
voicemail notification restriction msg-notification
voicemail operator telephone 0
voicemail live-record beep duration 1000
voicemail live-record pilot-number 6005
voicemail mailbox owner "Fax" size 420
description "Fax mailbox"
end mailbox

voicemail mailbox owner "dn6700" size 420
description "Thomatis mailbox"
messagesize 60
end mailbox

voicemail mailbox owner "mthomati" size 420
description "mthomati mailbox"
messagesize 60
end mailbox

voicemail mailbox owner "routehub" size 500
description "Routehub mailbox"
messagesize 60
end mailbox

voicemail mailbox owner "ucm6_mthomati" size 100

description "ucm6_mthomati mailbox"
messagesize 60
end mailbox

voicemail mailbox owner "ucm6_sip1" size 100
description "ucm6_sip1 mailbox"
messagesize 60
end mailbox

voicemail mailbox owner "ucm6_routehub" size 100
description "ucm6_routehub mailbox"
messagesize 60
end mailbox

voicemail notification owner dn6700 enable
voicemail notification owner routehub enable
voicemail notification owner Fax enable

Vuc SMB CP Cme

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RouteHub Group, LLC Page 1 www.routehub.

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