VUC-SMB-CP-CME

5/24/2018

VUC-SMB-CP-CME - slide pdf.c om

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

RouteHub Group, LLC

http://slide pdf.c om/re a de r/full/vuc -smb-c p-c me

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Table of Contents 1

Introduction

8

2

Concepts

9

3

Design 3.1 Our Design Small Business Voice Design 3.2 Requirements 3.3 Solutions and Topology 3.4 Topology Services and Sub-Services 3.5 Hardware & Software 3.6 Network Diagram

11 11 13 13 14 15 16

4

Configuration for CME 4.1 Initial Configuration 4.2 CallManager Express

17 17 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 4.4 4.5

Voice and Data VLAN Configuration Configuring DHCP on Cisco IOS Mapping an analog line (DID) to an IP phone

25 27 27

4.6 4.7 4.8 4.9 4.10 4.11 4.12 4.13

Configuring FXS port as a SCCP port CME as SIP Server for SIP Clients Blocking incoming calls from PSTN Setting up an Authenticated SIP Trunk to SIP Provider Phone Directory Single Number Reach (SNR) Sending Calls to Voicemail (CUE) Conferencing

30 32 33 35 38 38 39 41

4.13.1

41

MeetMe Conferencing

4.13.2 Ad-Hoc Conferencing

4.14 4.15 4.16 4.17 4.18 4.19 4.20 4.21

44

Paging Personal Speed Dial Upgrading CallManager Express Intercom Hunt Group Call Park How to setup Phone Softkey templates Extension Mobility

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4.22 4.23 4.24 4.25 4.26

4.27 4.28 4.29 4.30 4.31

How to setup a custom ring tone? Call Center Fax to Email using T.37 Phone Services Cisco CME using Exchange 2007 UM

52 55 58 61 62

Using a XML Menu File for Phone Services Installing SIP Firmware on Cisco 7940/7960 VoiceView Installing SIP Firmware on Cisco 7941/7961/7962+ Cisco Unified CallConnector

64 66 71 80 84

4.31.1

Server Installation

4.31.2

Components

85 109

4.31.3 Adding a new user

5

111

4.31.4

Client Installation

117

4.31.5

Using Cisco Unified CallConnector

124

134 134

Configuration for CUE 5.1 Access to CUE 5.1.1 5.1.2

5.2

134

CME Configuration Console into the CUE Service Engine.

136

136

Unity Express

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 5.4 5.5 5.6 5.7

Upgrade CUE to Version 7.x Coping Files to CUE via CLI Auto Attendant Voicemail Email Notifications CUE and CME on separate routers

5.7.1

142 145 146 149 151 151

CME Router

5.7.2

CUE Router (Cisco CUE Router Configuration)

154

5.7.3

CUE Router (CUE Configuration)

155

5.8 5.9

6

158 160

Live Record Downgrade CUE software

5.10 Basic CUE Start-Up Wizard

167

Monitor 6.1 Operations

170 170

6.1.1

IP Phones

170

6.1.2

Conferencing and DSP resources

172

6.1.3

Dial Plan and Cisco CallManager Express

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6.1.4 6.1.5

External Calling summary

191

6.1.6

Email Notification and Voice Messaging (CUE)

192

6.2

194

Troubleshooting

6.2.1

6.2.2 6.2.3

7

182

SIP

Root Causes

194

Initial questions to ask Typical fixes

194 195

196 196

Sample Full Configuration 7.1 CME and CUE on UC520 7.1.1

CME 7.1 on UC520

196

7.1.2

CUE 7.0.1 on UC520

216

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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 k now how to manage a network in real time.

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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 aagain, business. Well because of those capabilities a high price tag networks is associated with it. Then you get what you pay for. I have been deploying Cisco 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 RouteHub Group, LLC


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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.

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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 be sent via email. Other unique features with the voice solution would also bemessages great, but can 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. 



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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!      

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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 it’s 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

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Once the solutions have been determined it is time to build our topology. The topology is basically the framework in our design that doesn’t 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 RouteHub Group, LLC


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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.

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

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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.

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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. 2.

3.

We will need to configure our CME server, which is done under “telephony-service”. 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 “ephonedn”. 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 no no no

address 10.67.78.1 255.255.255.0 ip redirects ip unreachables ip proxy-arp

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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: B ANNER 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: RouteHub Group, LLC


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1 2 3 4 5 6

Dateline Standard Time -720 Samoa Standard Time -660 Hawaiian Standard Time -600 Alaskan Standard/Daylight Time -540 Pacific Standard/Daylight Time -480 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 GMT Azores Standard/Daylight Time 21 Standard/Daylight Time +0-60 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 48 49 50 51 52 53

E. Australia Standard Time +600 AUS Eastern Standard/Daylight Time +600 West Pacific Standard Time +600 Tasmania Standard/Daylight Time +600 Central Pacific Standard Time +660 Fiji Standard Time +720 New Zealand Standard/Daylight Time +720

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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: W EB 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

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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 7961GE 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

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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:

ephone-dn 4 dual-line ring internal number 1002 label 1002 (Family Room) name 1002 call-forward busy 6000 call-forward noan 6000 timeout 15 For DN 6700, we will add a description that would display the external number right above our lines/extensions on our IP phone:

ephone-dn 10 dual-line number 6700 label 6700 (Main) description 9252302203 name 6700 call-forward busy 6000 call-forward noan 6000 timeout 15 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

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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 don’t need to configure that.

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For button 1 on our IP phone it will use DN profile 1, which is for 6700. For button 2 on our IP phone it will use DN profile 3, which is for 1001. 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 don’t need to configure that.

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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. For button 2 on our IP phone it will use DN profile 4, which is for 1002.

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.

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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 access vlan 10 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

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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

interface FastEthernet0/1 description EDGE: VLAN DATA+VOICE switchport access vlan 10 switchport mode access switchport voice vlan 100 spanning-tree portfast interface GigabitEthernet0/1 description UPLINK: LAN CORE OR AGG switchport trunk allowed vlan 10,100 switchport mode trunk 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

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4.4 Configuring DHCP on Cisco IOS To enable DHCP services on our network this can be configured on a Cisco router, L3switch, or firewall, but it’s 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, let’s explain what we are talking about. Let’s 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/ voice translation-rule 2 rule 1 /^9/ /29/

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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 Let’s 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 dial-peer voice 2 voip 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 dial-peer voice 29 pots destination-pattern 29T port 1/0/1

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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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 202 caller-id enable

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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 let’s 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 switchback method 5 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 callerID on this port.

dial-peer voice 14 pots service stcapp port 0/0/0 voice-port 0/0/0 caller-id enable

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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.

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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, let’s 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 voice register pool 2 id mac 0019.D111.D2E8 number 1 dn 2 username 8778 password cisco6778 codec g711ulaw

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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:

voice translation-rule 5 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.

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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 w hen 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 r outer will match this dial peer because of the syntax “incoming called-number .”

dial-peer voice 100 pots 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.

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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 provider ’s control panel. We also need to specify the SIP provider’s server IP address or host/domain name.

sip-ua authentication username 19252302203 password cisco6778 no remote-party-id retry invite 2 10 retry register 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 voiph323 to h323 allow-connections allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 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 it’s full DID number would be 925-230-2203, our dedicated SIP number.

ephone-dn 13 dual-line number 7700 secondary 19252302203 name 7700 call-forward busy 6000 call-forward noan 6000 timeout 15

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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.

voice translation-rule 2 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.

voice translation-rule 3 rule 1 /^.*/ /19252302203/ voice translation-profile routehub-tp-sip-outgoing 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 byour a pattern numbers that match anywe local or long “session distancetarget call. This would be routed across authenticated SIP trunk (since specified sip -server”). We will also include codecs and DTMF details.

voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 dial-peer voice 12 voip 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

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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:

ephone-dn 3 dual-line number 1001 no-reg primary ephone-dn 4 dual-line number 1002 no-reg primary ephone-dn 10 dual-line number 6700 no-reg primary

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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)

ephone-dn 4 dual-line number 1002 no-reg primary 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.

snr 819252302203 delay 2 timeout 30 cfwd-noan 6000

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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.

dial-peer voice 600 voip destination-pattern 6000 session protocol sipv2 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.

ephone-dn 10 dual-line number 6700 no-reg primary call-forward busy 6000 call-forward noan 6000 timeout 15

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.

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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 number 8001.... no-reg primary mwi off

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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 it’s 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 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: 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 frequency conference 900 900 cadence 150 50 150 50 voice class custom-cptone routehub-join dualtone conference frequency 1200 1200 cadence 150 50 150 50

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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.

ephone-dn 22 dual-line number 6999 conference meetme no huntstop ephone-dn 23 dual-line number 6999 conference meetme preference 1 no huntstop

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ephone-dn 24 dual-line number 6999 conference meetme preference 2 no huntstop ephone-dn 25 dual-line 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 nohow MeetMe softkey existSoftkey then follow the section within this document discussing to setup a Phone 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.

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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 adhoc across two ephone-dn profiles for load distribution.

ephone-dn 26 dual-line number 6998 name Conference conference ad-hoc preference 1 no huntstop ephone-dn 27 dual-line 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. Let’s 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.

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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 P AGING First we need to configure Paging and assign what the directory number will be. also be configured or enabled for multicast.

This can

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 P AGING 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 P AGING 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.

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4.15 Personal Speed Dial To setup a personal speed dial or fast dial for a user’s 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.

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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 number A5002 no-reg primary 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 doesn’t require someone to answer the Intercom call. Remember it’s a like broadcast type service.

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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 don’t 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.

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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

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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 voice user-profile 2 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 let’s 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 r outer. Let’s say user1 is at user2’s desk. User1 can login to user2’s 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 number 6700,A5001,7700,1001 type feature-ring voice logout-profile 2 pin 6700 user 16700 password 6700 number 6700,A5002,1002 type feature-ring

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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 initially IP address is the on our CME router that is associated for our CME configured under our IP 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”.

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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:

24 24.raw 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:

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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.

ephone-dn 10 dual-line ring external number 6700 no-reg primary

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

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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.

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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.

ephone-dn 14 dual-line number 2001 ephone-dn 15 dual-line number 2002

STEP 2: HUNT GROUP Next we will configure a hunt thattimeout will include the two DN we each configured from step The pilot number will be 6721group and the for ringing between number listed in 1. 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: C ALL 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

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STEP 4: DOWNLOAD C ALL 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 router’s flash successfully before continuing.

STEP 5: CONFIGURE C ALL 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 RouteHub Group, LLC


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param param param param

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

service queue flash:app-b-acd-2.1.2.3.tcl param queue-len 5 param queue-manager-debugs 1 param aa-hunt2 6721 param number-of-hunt-grps 1

STEP 6: DIAL PEER ASSOCIATED WITH C ALL 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.

dial-peer voice 1009 voip service aa destination-pattern 6720 session target ipv4:10.67.78.1 incoming called-number 6720 dtmf-relay h245-alphanumeric codec g711ulaw no vad

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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 router’s flash successfully before continuing.

STEP 2: ENABLE TCP SCRIPT APPLICATIONS Next, once the TCP scripts have been copied to the router’s 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

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STEP 3: CONFIGURE F AX 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 mta mta mta mta mta mta mta mta mta

send with-subject both send postmaster [email protected] send mail-from hostname routehub.com send mail-from username IncomingFax send return-receipt-to hostname routehub.com send return-receipt-to username ROUTEHUB receive aliases routehub.com receive aliases 10.67.78.6 receive maximum-recipients 10 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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 6700 caller-id enable

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STEP 5: F AX 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 it’s 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.

dial-peer voice 100 pots service fax_detect destination-pattern 9.T incoming called-number 6700 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. Let’s 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 destination-pattern 6700 information-type fax session target mailto:[email protected]

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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 won’t 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 connection plar opx 6700 caller-id enable dial-peer voice 101 pots service onramp incoming called-number 6700 direct-inward-dial port 0/1/1 dial-peer voice 7 mmoip description FAX service fax_on_vfc_onramp_app out-bound destination-pattern 6700 information-type fax session target mailto:[email protected]

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.

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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 allow-connections sip to sip supplementary-service h450.12 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.

dial-peer voice 303 voip description EXCH2007-UM destination-pattern 671. session protocol sipv2 session target ipv4:10.67.78.92 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

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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. uc01tra#show sip-ua calls SIP UAC CALL INFO Call 1 SIP Call ID : [email protected] 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: Destination Name : Number of Media Streams : Number of Active Streams: RTP Fork Object : Media Mode : Media Stream 1 State of the stream Stream Call ID Stream Type Stream Media Addr Type Negotiated Codec Codec Payload Type Negotiated Dtmf-relay Dtmf-relay Payload Type

[10.67.78.92]:5065 10.67.78.92 1 1 0x0 flow-through : : : : : : : :

STREAM_ACTIVE 1172704 voice+dtmf (1) 1 g711alaw (160 bytes) 8 rtp-nte 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 Number of SIP User Agent Client(UAC) calls: 1 SIP UAS CALL INFO Number of SIP User Agent Server(UAS) calls: 0

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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 potential we do then? weupload need to create XMLthen file that will include two and otherSimple, services, this file toaacustom web server point the phonethese 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 Phone Services Please make your selection. VoiceView http://10.67.5.2/voiceview/common/login.do Weather,News,Stocks http://phone-xml.berbee.com/menu.xml

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 don’t 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

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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.

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4.28 Installing SIP Firmware on Cisco 7940/7960



1.

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

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.

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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:


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5.

We need to create a new file that is called "SIPmacaddress.cnf" where "macaddress" 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 callerID to be "SIP User (8778).


6.

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

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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 default-router 10.67.99.1 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: interface FastEthernet0/1/6 switchport access vlan 11

10. Below is a summary of the SIP server configuration on our Cisco CME router. voice service voip allow-connections sip to sip 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 voice register dn 2 number 8778 name ROUTEHUB User (SIP) voice register pool 1 id mac 0012.00A7.72EA number 1 dn 2 username 8778 password cisco6778 codec g711ulaw

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

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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:

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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:



2.

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

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 url authentication http://10.67.5.2/voiceview/authentication/authenticate.do

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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. Phone Services Please make your selection. VoiceView http://10.67.5.2/voiceview/common/login.do Weather,News,Stocks http://phone-xml.berbee.com/menu.xml?opts=95377

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

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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"

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

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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”).

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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.

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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:

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

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

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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 noWe big will concern thatthis is the biggest reason why the SIP firmware installation doesn't work. discuss 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:

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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.

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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:

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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 default-router 10.67.99.1 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.

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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 user’s 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.

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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 let’s 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

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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.

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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:

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During the installation it will ask about the Licensing Option. We can input the PAKID 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 it’s installed on a non-Cisco approved system.

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

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5.

Once the installation is completed it will start the Step Wizard for us to complete (or confirm) the following tasks. It’s 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.

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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.

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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.

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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”.

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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.

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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.

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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”.

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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.

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

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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”.

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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”. User s 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”.

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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.

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

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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.

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

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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.

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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.

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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.

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4.31.2

Components

Once UCC is installed and configured it’s 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:

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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.

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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.

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2.

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

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3.

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

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4.

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

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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”:

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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.

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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.

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3.

Now the installation will continue.

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4.

On the next page we will specify our customer information

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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.

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6.

From here we will continue with the installation process.

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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.

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8.

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

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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.

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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.

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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.

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5.

Let’s 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”.

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6.

Now let’s 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”

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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”.

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9.

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

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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 let’s 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.

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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.

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

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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 it’s 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 N AME 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 FastEthernet0/0 1.1.1.73 In0/0 unassigned FastEthernet0/1/0 unassigned

OK? YES YES YES

Method NVRAM TFTP unset

Status up up up

Protocol up up up

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

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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:

interface Integrated-Service-Engine0/0 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

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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 [email protected] -----------------------------------------------------------------------

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. 3. 4.

Enable Voicemail Services including MWI Create User Voice Mailboxes Enable Other CUE Services like Auto Attendant

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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 let’s 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

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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 C ALL H ANDLING (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 user’s 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

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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.

dial-peer voice 600 voip destination-pattern 6000 session protocol sipv2 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.


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.

ephone-dn 10 dual-line number 6700 no-reg primary call-forward busy 6000 call-forward noan 6000 timeout 15

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 RouteHub Group, LLC


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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 number 8001.... no-reg primary 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"

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STEP 4: CREATE USER VOICE M AILBOX 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 voicemail voicemail voicemail

callerid default language en_US default mailboxsize 420 broadcast recording time 300

voicemail default messagesize 240 voicemail notification restriction msg-notification voicemail operator telephone 0

STEP 6: CONFIGURE FULL N AME TO SUPPORT DIAL-BY-N AME 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

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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 onthe-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.

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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.

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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:: WARNING:: WARNING:: Would you

This command will install the necessary software to complete an upgrade. It is recommended that a backup be done before installing software. 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-vmlicense_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

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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

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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 doesn’t 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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect 2 timeouts wait-release 2 connection enable plar 6003 caller-id 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.

dial-peer voice 600 voip destination-pattern 6... session protocol sipv2 session target ipv4:10.67.78.2 dtmf-relay sip-notify 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... || || || || |||| |||| ..:||||||:..:||||||:.. c i s c o S y s t e m s OFFICAL USE ONLY! RouteHub Group, LLC (925) 230-2203 www.routehub.com [email protected] RouteHub Group, LLC


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

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 C ALL H ANDLE/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.

ccn trigger sip phonenumber 6003 application "autoattendant" enabled locale "en_US" maxsessions 4 end trigger

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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. It’s 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 ccn trigger sip phonenumber 6006 application "promptmgmt" enabled idletimeout 5000 locale "en_US" maxsessions 1 end trigger

STEP 7: USING THE PROMPT M ANAGER 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, let’s say we dial 6006. It will ask for us to enter our extension so let’s 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 t hen active it. That’s 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.

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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 phone’s 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 [email protected].

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.


notification notification notification notification

enable preference all allow-login 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 [email protected]. voicemail configuration outgoing-email from-address [email protected]

TEP

NABLE

OICEMAIL NOTIFICATION FOR USER WITH NUMBER

S 4: Eneed toVenable voicemail notification for the user or users 6700 Next we 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 create username dn6700 phonenumber “6700”

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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 [email protected]) 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 username username username username username username username username username username

dn6700 dn6700 dn6700 dn6700 dn6700 dn6700 dn6700 dn6700 dn6700 dn6700 dn6700

profile profile profile profile profile profile profile profile profile profile profile

VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700 VM-6700

email email email email email email email email email email email

address enable [email protected] preference all attach schedule day 1 active from schedule day 2 active from schedule day 3 active from schedule day 4 active from schedule day 5 active from schedule day 6 active from schedule day 7 active from

01:00 01:00 01:00 01:00 01:00 01:00 01:00

to to to to to to to

24:00 24:00 24:00 24:00 24:00 24:00 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  

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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, let’s 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 let’s specify all SIP and H.323 communications are allowed to/from our CME router.

voice service voip allow-connections h323 to h323 allow-connections h323 toh323 sip allow-connections sip to allow-connections sip to sip

STEP 2: VOIP DIAL PEER FOR CUE SERVICES (VM, AA, PM) Next let’s 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).


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.

dial-peer voice 991 voip session protocol sipv2 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

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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 call-forward busy 2000 call-forward noan 2000 timeout 15 ephone 24 device-security-mode none 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.


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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect timeouts wait-release 2 2 connection plar 2001 caller-id enable

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STEP A: SUMMARY CONFIGURATION

voice service voip allow-connections allow-connections allow-connections allow-connections

h323 to h323 h323 to sip sip to h323 sip to sip

dial-peer voice 2000 voip destination-pattern 2... session protocol sipv2 session target ipv4:192.168.1.2 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 991 voip session protocol sipv2 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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect 2 timeouts wait-release 2 connection plar 2001 caller-id enable ephone-dn 24 number 199 label testuser - 199 name testuser call-forward busy 2000 call-forward noan 2000 timeout 15 ephone 24 device-security-mode none username "testuser" mac-address 0001.4A25.68E0 type CIPC button 1:24

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

dial-peer voice 100 voip destination-pattern 1.. session protocol sipv2 session target ipv4:192.168.3.254 dtmf-relay sip-notify codec g711ulaw no vad

STEP 2: VOIP DIAL PEER TO CUE Next let’s 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.


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.

dial-peer voice 101 voip destination-pattern 80[0,1]... session protocol sipv2 session target ipv4:192.168.3.254 dtmf-relay sip-notify codec g711ulaw no vad

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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 gateway address "192.168.1.1" 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.

ccn trigger sip phonenumber 2000 application "voicemail" enabled maxsessions 6 end trigger ccn trigger sip phonenumber 2001 application "autoattendant" enabled locale "en_US" maxsessions 6 end trigger ccn trigger sip phonenumber 2002 application "promptmgmt" 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.

ccn application ciscomwiapplication description "ciscomwiapplication" enabled maxsessions 4 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "801" parameter "strMWI_ON_DN" "800" end application

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

groupname Users create 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 dial-peer voice 100 voip destination-pattern 1.. session protocol sipv2 session target ipv4:192.168.3.254 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 2000 voip destination-pattern 2... session protocol sipv2 session target ipv4:192.168.1.2 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 101 voip destination-pattern 80[0,1]... session protocol sipv2 session target ipv4:192.168.3.254 dtmf-relay sip-notify codec g711ulaw no vad

-------------------------username U199 create username U199 phonenumber "199" groupname Users create ccn application ciscomwiapplication description "ciscomwiapplication" enabled maxsessions 4 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "801" parameter "strMWI_ON_DN" "800" end application

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ccn subsystem sip gateway address "192.168.1.1" end subsystem ccn trigger sip phonenumber 2000 application "voicemail" enabled maxsessions 6 end trigger ccn trigger sip phonenumber 2001 application "autoattendant" enabled locale "en_US" maxsessions 6 end trigger ccn trigger sip phonenumber 2002 application "promptmgmt" 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

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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 subscriber’s 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 call-forward all 6000

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dial-peer voice 600 voip destination-pattern 6... session protocol sipv2 session target ipv4:10.67.78.2 dtmf-relay sip-notify 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.

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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 subscriber’s 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 whenlicenses we look available. at the licensing we see that this voicemail is has disabled and that there aredevice, no mailbox Therefore, doing upgrade 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

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WARNING:: This command will download the necessary software to WARNING:: complete a clean install. It is recommended that a backup be done WARNING:: before installing software. Would you like to continue?[confirm] Downloading ftp cue-vm-k9.nme.7.0.1.pkg Bytes downloaded :

182508

Validating package signature ... done - Parsing package manifest files... complete. Validating installed manifests ............complete. - Checking Package dependencies... complete. Downloading ftp cue-vm-langpack.nme.7.0.1.pkg Bytes downloaded : 1096191 Validating package signature ... done Found Add-On Subsystem SID: e2e81cc6-39b5-47e1-9f83-b83c897fc50c Name: CUE Voicemail Language Support Version: 7.0.1.0 .... - Parsing package manifest files... complete. - Checking Package dependencies... complete. - 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) RouteHub Group, LLC


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14 15 16 17 18 19 20 21

ENG DEU DAN PTB KOR CHS JPN RUS

CUE CUE CUE CUE CUE CUE CUE CUE

Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail

UK English (7.0.1.0) German (7.0.1.0) Danish (7.0.1.0) Brazilian Portuguese (7.0.1.0) Korean (7.0.1.0) Mandarin Chinese (7.0.1.0) Japanese (7.0.1.0) 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 Language Installation Menu: # Selected SKU Language Name (version) ---------------------------------------------------------------------1 ITA CUE Voicemail Italian (7.0.1.0) 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16

*

ESP ENU FRA ESO ESM ARA NLD SVE NOR FRC TUR HUN ENG DEU DAN

CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE

Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail

European Spanish (7.0.1.0) US English (7.0.1.0) European French (7.0.1.0) Latin American Spanish (7.0.1.0) Mexican Spanish (7.0.1.0) Arabic (7.0.1.0) Dutch (7.0.1.0) Swedish (7.0.1.0) Norwegian (7.0.1.0) Canadian French (7.0.1.0) Turkish (7.0.1.0) Hungarian (7.0.1.0) UK English (7.0.1.0) German (7.0.1.0) 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:x ui_install scripts executed successfully.

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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 WARNING:: before installing software. Would you like to continue?[confirm] Validating package signature ... done - Parsing package manifest files... complete. Validating installed manifests ............complete. - Checking Package dependencies... complete. Validating package signature ... done 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: 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 5 6 7 8 9 10 11 12 13 14

FRA ESO ESM ARA NLD SVE NOR FRC TUR HUN ENG

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CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE

Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail

European French (7.0.1.0) Latin American Spanish (7.0.1.0) Mexican Spanish (7.0.1.0) Arabic (7.0.1.0) Dutch (7.0.1.0) Swedish (7.0.1.0) Norwegian (7.0.1.0) Canadian French (7.0.1.0) Turkish (7.0.1.0) Hungarian (7.0.1.0) UK English (7.0.1.0)

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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 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 4 5 6 7 8 9 10 11 12 13 14 15 16 17

*

ENU FRA ESO ESM ARA NLD SVE NOR FRC TUR HUN ENG DEU DAN PTB

CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE CUE

Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail Voicemail

US English (7.0.1.0) European French (7.0.1.0) Latin American Spanish (7.0.1.0) Mexican Spanish (7.0.1.0) Arabic (7.0.1.0) Dutch (7.0.1.0) Swedish (7.0.1.0) Norwegian (7.0.1.0) Canadian French (7.0.1.0) Turkish (7.0.1.0) Hungarian (7.0.1.0) UK English (7.0.1.0) German (7.0.1.0) Danish (7.0.1.0) 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: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

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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 W IZARD 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:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT:: IMPORTANT::

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

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

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

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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

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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 N AME Enter Domain Name (mydomain.com, or enter to use localdomain): routehub.com

STEP 3: DNS IMPORTANT:: DNS Configuration: IMPORTANT:: IMPORTANT:: instead IMPORTANT:: In order IMPORTANT:: your IMPORTANT::

This allows the entry of hostnames, for example foo.cisco.com, of IP addresses like 1.100.10.205 for application configuration. to set up DNS you must know the IP address of at least one of 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 the Secondary DNS Server (other than Primary) (IP address, orof enter to bypass):

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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 RouteHub Group, LLC


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14) 15) 16) 17) 18) 19) 20) 21)

Central Time - Michigan - Dickinson, Gogebic, Iron & Menominee Counties Central Time - North Dakota - Oliver County Central Time - North Dakota - Morton County (except Mandan area) Mountain Time Mountain Time - south Idaho & east Oregon Mountain Time - Navajo Mountain Standard Time - Arizona 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: Universal Time is now:

Thu Mar 18 02:35:33 PDT 2010. 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

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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 7905 7906 7910 7911 7912 7914 7920 7921 7931 7935 7936 7940 7941 7941GE 7960 7961

7902 phone status 7905 phone status 7906 phone status 7910 phone status 7911 phone status 7912 phone status 7914 phone status 7920 phone status 7921 phone status 7931 phone status 7935 phone status 7936 phone status 7940 phone status 7941 phone status 7941GE phone status 7960 phone status 7961 phone status

7961GE 7970 7971 7985 H.H.H anl ata attempted-registrations bri cfa dn dnd login offhook overlay phone-load registered

7961GE phone status 7970 phone status 7971 phone status 7985 phone status mac address ANL port status ata phone status Attempted ephone list BRI port status registered ephones with call-forward-all set Dn with tag assigned registered ephones with do-not-disturb set phone login status Offhook phone status registered ephones with overlay DNs Ephone phoneload information Registered ephone status

remote ringing sockets summary tapiclients telephone-number unregistered |

non-local phones (with no arp entry) Ringing phone status Active ephone sockets Summary of all ephone Ephone status of tapi client Telephone number assigned Unregistered ephone status Output modifiers

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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 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0 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 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 debug:0 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 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0 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 ephone’s 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 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7 IP:10.67.78.31 49949 7970 keepalive 3689 max_line 8 button CH2 IDLE IDLE shared button 1: 2: dn dn 10 11 number number 6700 A5001CH1 autoCONNECTED dial A5002 CH1 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 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:6 IP:10.67.78.111 1050 7920 keepalive 53051 max_line 6

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button 1: dn 10 number 6700 CH1 CONNECTED CH2 button 2: dn 12 number A5002 auto dial A5001 CH1

IDLE IDLE

shared 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 DN 10 chan 1 incoming 292 answered 138 outgoing 508 answered 317 busy 3 Far-end disconnect at: connect 61 alert 29 hold 0 ring 154 Last 64 far-end disconnect cause codes 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 DN 10 chan 2 incoming 1 answered 1 outgoing 2 answered 1 busy 0 Far-end disconnect at: connect 0 alert 0 hold 0 ring 0 Last 640 far-end 0 28 0 0 0 0 0 disconnect 0 0 0 0 0 0cause 0 0 codes 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 Call Ref 884 called 106778 calling 6700 Total Tx Pkts 837 bytes 143964 Rx Pkts 777 bytes 133644 Lost 22 Final Jitter 24 Latency 373 Lost 0 Signal Level to phone 0 (-78 dB) peak 0 (-78 dB) Packets counted by router 778

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DN 11 chan 1 incoming 5 answered 5 outgoing 3 answered 3 busy 0 Far-end disconnect at: connect 0 alert 0 hold 0 ring 0 Last 64 far-end disconnect cause codes 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 DN 12 chan 1 incoming 3 answered 3 outgoing 5 answered 5 busy 0 Far-end disconnect at: connect 0 alert 0 hold 0 ring 0 Last 64 far-end disconnect cause codes 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 Total Tx Pkts 91 bytes 15652 Rx Pkts 92 bytes 15824 Lost 0 Final Jitter 0 Latency 0 Lost 0 Signal Level to phone 0 (-78 dB) peak 0 (-78 dB) Packets counted by router 0 DN 13 chan 1 incoming 50 answered 5 outgoing 29 answered 15 busy 0 Far-end disconnect at: connect 3 alert Last 64 far-end disconnect cause codes 19 16 16 16 16 19 16 19 16 16 16 16 16 16 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 1 1 1 19 1 19 0 0 0 0 0 0 0 0 0 0

6 hold 0 ring 45 16 16 19 19 19 19 19 1 1

DN 13 chan 1 (7700) voice quality statistics for last call Call Ref 897 called 7700 calling 6700 Total Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0 Final Jitter 31 Latency 790 Lost 0 Signal Level to phone 0 (-78 dB) peak 0 (-78 dB) Packets counted by router 0 DN 13 chan 2 incoming 0 answered 0 outgoing 2 answered 0 busy 0 Far-end disconnect at: connect 0 alert 0 hold 0 ring 0 Last 64 far-end disconnect cause codes 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 DN 13 chan 2 (7700) voice quality statistics for last call Call Ref 868 called calling Total Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0 Final Jitter 32 Latency 446 Lost 0 Signal Level to phone 0 (-78 dB) peak 0 (-78 dB) Packets counted by router 0

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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 PORT CH ======== == 50/0/10 1 EFXS_CONNECT

ephone-dn summary DN STATE MWI_STATE ======== ========= CONNECTED NONE

CODEC VAD VTSP STATE ===== === =========== g711ulaw n S_CONNECT

VPM STATE =========

50/0/10 50/0/20 50/0/21 50/0/11 50/0/12 50/0/13 50/0/13 50/0/1 50/0/2 50/0/22 50/0/22 50/0/23 50/0/23 50/0/24 50/0/24 50/0/25 50/0/25

IDLE IDLE IDLE IDLE IDLE IDLE IDLE IDLE IDLE CONNECTED IDLE IDLE IDLE IDLE IDLE IDLE IDLE

g711ulaw -

EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_CONNECT EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK

2 1 1 1 1 1 2 1 1 1 2 1 2 1 2 1 2

NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE NONE

n -

S_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 TYPE ==== edsp edsp edsp edsp edsp edsp edsp edsp edsp

DSP NUM === 001 002 003 004 005 006 007 008 009

CH == 01 01 01 02 01 01 01 02 01

DSPWARE CURR BOOT PAK TX/RX CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT ======== ========== ===== ======= === == ========= == ===== ============ g711ulaw 0.1 IDLE 50/0/1.1 g729r8 p 0.1 IDLE 50/0/2.1 g711ulaw 0.1 IDLE 50/0/10.1 g711ulaw 0.1 IDLE 50/0/10.2 g711ulaw 0.1 IDLE 50/0/11.1 g711ulaw 0.1 IDLE 50/0/12.1 g729r8 0.1 IDLE 50/0/13.1 g711ulaw 0.1 IDLE 50/0/13.2 g729r8 p 0.1 IDLE 50/0/20.1

edsp edsp edsp edsp edsp edsp edsp edsp edsp

010 011 012 013 014 015 016 017 018

01 01 02 01 02 01 02 01 02

g729r8 p g711ulaw g711ulaw g729r8 p g729r8 p g729r8 p g729r8 p g729r8 p g729r8 p

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0.1 0.1 0.1 0.1 0.1 0.1 0.1 0.1 0.1

IDLE IDLE IDLE IDLE IDLE IDLE IDLE IDLE IDLE

50/0/21.1 50/0/22.1 50/0/22.2 50/0/23.1 50/0/23.2 50/0/24.1 50/0/24.2 50/0/25.1 50/0/25.2

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----------------------------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 DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ===== === == ========= ========== ===== ======= === == ========= == ==== ============ *DSP SIGNALING CHANNELS* DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ===== === == ========= ========== ===== ======= === == ========= == ==== ============ 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 DSP TYPE ==== edsp edsp edsp edsp edsp edsp edsp edsp edsp edsp edsp edsp

DSP NUM === 001 002 003 004 005 006 007 008 009 010 011 012

CH == 01 01 01 02 01 01 01 02 01 01 01 02

DSPWARE CURR BOOT PAK TX/RX CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT ======== ========== ===== ======= === == ========= == ===== ============ g711ulaw 0.1 IDLE 50/0/1.1 g729r8 p 0.1 IDLE 50/0/2.1 g711ulaw 0.1 IDLE 50/0/10.1 g711ulaw 0.1 IDLE 50/0/10.2 g711ulaw 0.1 IDLE 50/0/11.1 g711ulaw 0.1 IDLE 50/0/12.1 g729r8 0.1 IDLE 50/0/13.1 g711ulaw 0.1 IDLE 50/0/13.2 g729r8 p 0.1 IDLE 50/0/20.1 g729r8 p 0.1 IDLE 50/0/21.1 g711ulaw 0.1 IDLE 50/0/22.1 g711ulaw 0.1 IDLE 50/0/22.2

edsp edsp edsp edsp edsp

013 014 015 016 017 018

01 02 01 02 01 02

g729r8 g729r8 g729r8 g729r8 g729r8

p p p p p

0.1 0.1 0.1 0.1 0.1

IDLE IDLE IDLE IDLE IDLE

50/0/23.1 50/0/23.2 50/0/24.1 50/0/24.2 50/0/25.1 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 DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ===== === == ========= ========== ===== ======= === == ========= == ==== ============ C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510

001 001 001 001 001 001 001 001 001 001 001 001

01 02 03 04 05 06 07 08 09 10 11 12

None None None None None None None None None None None None

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21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0 21.4.0

idle idle idle idle idle idle idle idle idle idle idle idle

idle idle idle idle idle idle idle idle idle idle idle idle

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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

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C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510 C5510

001 001 001 001 002 002 002 002 002

13 14 15 16 01 02 03 04 05

None None None None None None None None None

21.4.0 21.4.0 21.4.0 21.4.0 1.1.137 1.1.137 1.1.137 1.1.137 1.1.137

idle idle idle idle idle idle idle idle idle

idle idle idle idle idle idle idle idle idle

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

C5510 1.1.137 0 0 C5510 002 002 06 07 None None 1.1.137 idle idle idle idle 0 0 0 0 C5510 002 08 None 1.1.137 idle idle 0 0 0 C5510 002 09 None 1.1.137 idle idle 0 0 0 C5510 002 10 None 1.1.137 idle idle 0 0 0 C5510 002 11 None 1.1.137 idle idle 0 0 0 C5510 002 12 None 1.1.137 idle idle 0 0 0 C5510 002 13 None 1.1.137 idle idle 0 0 0 C5510 002 14 None 1.1.137 idle idle 0 0 0 C5510 002 15 None 1.1.137 idle idle 0 0 0 C5510 002 16 None 1.1.137 idle idle 0 0 0 ------------------------END OF FLEX VOICE CARD 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

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

0 0

UP UP

conf conf

2 2

1.1.137 1.1.137

N/A N/A

FREE FREE

RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED 1 1

-

-

-

Total number of DSPFARM DSP channel(s) 2

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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 aretoother destination such asthen for SIP and our FXO Port (0/1/0), which is connected our PSTN. If weports dial “9” first a fulltrunks number it would match destination pattern “9.T” and r oute 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 100 pots up up 9.T 0 up 600 voip up up 6000 0 syst ipv4:192.168.5.2 20001 pots up up 6700$ 0 20002 pots up up 8000.... 0 20003 pots up up 8001.... 0 20004 pots up up A5001$ 0 20005 pots up up A5002$ 0 20006 pots up up 7700$ 0 20007 pots up up 19252302203$ 9 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 20009 pots up up 6002$ 0 20010 pots up up 6999$ 0 20011 pots up up 6999$ 1 20012 pots up up 6999$ 2 20013 pots up up 6999$ 3

PORT 0/1/0 50/0/10 50/0/20 50/0/21 50/0/11 50/0/12 50/0/13 50/0/13

50/0/1 50/0/2 50/0/22 50/0/23 50/0/24 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 source-address 10.67.78.1 port 2000 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 load load load load load load load load load load load

7914 S00104000100 7902 CP7902080001SCCP051117A 7906 SCCP11.8-0-3S 7911 SCCP11.8-0-3S 7921 CP7921G-1.0.1 7931 SCCP31.8-1-1SR2S 7936 cmterm_7936.3-3-5-0 7960-7940 P0030702T023 7941 TERM41.7-0-3-0S 7941GE TERM41.7-0-3-0S 7961 TERM41.7-0-3-0S 7961GE TERM41.7-0-3-0S

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load 7970 term70.default load 7971 TERM70.7-0-3-0S max-ephones 14 max-dn 56 max-conferences 8 gain -6 dspfarm units 3 dspfarm transcode sessions 0 dspfarm 1 mtp001d4567c690 dspfarm 2 dspfarm 3 hardware conference privacy no privacy-on-hold hunt-group report url suffix 0 to 200 hunt-group report every 2 hours # 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 network-locale[2] US network-locale[3] US network-locale[4] 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 moh music-on-hold.au time-format 12 date-format mm-dd-yy timezone 5 Pacific Standard/Daylight Time secondary-dialtone 9 url services http://10.67.78.2/voiceview/common/login.do url authentication http://10.67.78.2/voiceview/authentication/authenticate.do call-forward pattern .T call-forward system redirecting-expanded transfer-pattern 9.T 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 system message RouteHub UC520 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 create cnf-files version-stamp 7960 Mar 10 2009 14:54:25 transfer-system full-consult dss transfer-digit-collect new-call auto assign 10 to 19 local directory service: enabled. Extension-assigner tag-type ephone-tag.

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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 interdigit 5 timeout ringing 15 ! voice-port 50/0/20 station-id number 80000000 timeout interdigit 5 ! voice-port 50/0/21 station-id number 80010000 timeout interdigit 5 ! voice-port 50/0/11 station-id number A5001 station-id name Intercom timeout interdigit 5 ! voice-port 50/0/12 station-id number A5002 station-id name Intercom timeout interdigit 5 ! voice-port 50/0/13 station-id number 7700 station-id name 7700 timeout interdigit 5 timeout ringing 15 ! voice-port 50/0/1 station-id number 6001 station-id name Routehub Paging S timeout interdigit 5 ! voice-port 50/0/2 station-id number 6002 station-id name Routehub Call Par timeout interdigit 5 ! voice-port 50/0/22 station-id number 6999 timeout interdigit 5 ! voice-port 50/0/23 station-id number 6999 timeout interdigit 5 ! voice-port 50/0/24 station-id number 6999 timeout interdigit 5 ! voice-port 50/0/25 station-id number 6999 timeout interdigit 5

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

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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 dial-peer voice 20001 pots destination-pattern 6700$ no e164 registration huntstop call-forward busy 6000 call-forward noan 6000 progress_ind setup enable 3 port 50/0/10 dial-peer voice 20002 pots destination-pattern 8000.... no e164 registration huntstop progress_ind setup enable 3 port 50/0/20 dial-peer voice 20003 pots destination-pattern 8001.... no e164 registration huntstop progress_ind setup enable 3 port 50/0/21 dial-peer voice 20004 pots destination-pattern A5001$ no e164 registration huntstop progress_ind setup enable 3 port 50/0/11 dial-peer voice 20005 pots destination-pattern A5002$ no e164 registration huntstop progress_ind setup enable 3 port 50/0/12 dial-peer voice 20006 7700$ pots destination-pattern huntstop call-forward busy 6000 call-forward noan 6000 progress_ind setup enable 3 port 50/0/13 dial-peer voice 20007 pots preference 9 destination-pattern 19252302203$ huntstop call-forward busy 6000

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call-forward noan 6000 progress_ind setup enable 3 port 50/0/13 dial-peer voice 20008 pots destination-pattern 6001$ huntstop progress_ind setup enable 3 port 50/0/1 dial-peer voice 20009 pots destination-pattern 6002$ huntstop progress_ind setup enable 3 port 50/0/2 dial-peer voice 20010 pots destination-pattern 6999$ progress_ind setup enable 3 port 50/0/22 dial-peer voice 20011 pots preference 1 destination-pattern 6999$ progress_ind setup enable 3 port 50/0/23 dial-peer voice 20012 pots preference 2 destination-pattern 6999$ progress_ind setup enable 3 port 50/0/24 dial-peer voice 20013 pots preference 3 destination-pattern 6999$ progress_ind setup enable 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

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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 connections Display map Display min-se Display mwi Display register Display retry Display

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Active SIP Calls SIP Connections SIP status code to PSTN cause mapping table & vice versa Min-SE value SIP MWI server info SIP Register status SIP Protocol Retry Counts

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service statistics status timers

Display Display Display Display

SIP SIP SIP SIP

submode Shutdown status UA Statistics UA Listener Status 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 ============ 19252302203 6001 6002 6999 7700 8778 9.*

peer ============= 20007 20008 20009 20010 20006 40001 100

expires(sec) ============ 1299 105 132 141 105 105 159

registered =========== yes no no no no no 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. uc01tra#show sip-ua calls SIP UAC CALL INFO Call 1 SIP Call ID : [email protected] State of the call : STATE_ACTIVE (7) Substate of the call : SUBSTATE_NONE (0) Calling Number : 19252302203 Called Number : 12091117170 Bit Flags : 0xC04018 0x100 0x0 CC Call ID : 63803 Source IP Address (Sig ): 6.7.7.73 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 Streams : 1 Number of Active Streams: RTP Fork Object : 0x0 Media Mode : flow-through Media Stream 1 State of the stream : STREAM_ACTIVE Stream Call ID : 63803 Stream Type : voice+dtmf (1) Negotiated Codec : g729r8 (20 bytes) Codec Payload Type : 18 Negotiated Dtmf-relay : rtp-nte Dtmf-relay Payload Type : 101 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

Options-Ping ACTIVE:NO calls: 1 Number of SIPENABLED:NO User Agent Client(UAC) SIP UAS CALL INFO Number of SIP User Agent Server(UAS) calls: 0

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

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Total active connections No. of send failures No. of remote closures No. of conn. failures No. of inactive conn. ageouts

: : : : :

2 0 0 0 201

We can also get details of all active SIP connection if you are unsure what the established connections are. uc01tra#show connections Total active sip-ua connections : No. of send failures : No. of remote closures : No. of conn. failures : No. of inactive conn. ageouts :

udp detail 2 0 0 0 201

---------Printing Detailed Connection Report--------Note: ** Tuples with no matching socket entry - Do 'clear sip conn t ipv4::' to overcome this error condition ++ Tuples with mismatched address/port entry - Do 'clear sip conn t ipv4:: id ' 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

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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 Rule for Called number: 2 Rule for Redirect number: Rule for Redirect-target number: Translation Profile: RouteHub-tp-ucm6 Rule for Calling number: Rule for Called number: 10 Rule for Redirect number: Rule for Redirect-target 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 username 8701 password cisco6778 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:

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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. uc01tra#show voice register pool 2 Pool Tag 2 Config: Mac address is 0019.D111.D2E8 Number list 1 : DN 2 Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled username 8778 password cisco6778 service-control mechanism is not supported registration Call ID is d3ac1ab6-c5fa-1810-9921-0019d111d2e8@mat-dtop01tra active primary line is: 8778 contact IP address: 10.67.78.8 port 5061

Dialpeers created: Statistics: Active registrations

: 0

Total SIP phones registered: 0 Total Registration Statistics Registration requests : 14 Registration failed success Registration unRegister requests unRegister success unRegister failed

: : : : :

14 0 14 14 0

Now, this reflects what our SIP endpoint (configured to use number 8778) looks like when it is registered to our UC500 SIP proxy server. uc01tra#show voice register pool 2 Pool Tag 2 Config: Mac address is 0019.D111.D2E8 Number list 1 : DN 2 Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled username 8778 password cisco6778 service-control mechanism is not supported registration Call ID is 789d1de7-c8fa-1810-90c4-0019d111d2e8@mat-dtop01tra active primary line is: 8778 contact IP address: 10.67.78.8 port 5061

Dialpeers created:

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dial-peer voice 40001 voip destination-pattern 8778 session target ipv4:10.67.78.8:5061 session protocol sipv2 codec g711ulaw bytes 160 after-hours-exempt FALSE Statistics: Active registrations

: 1

Total SIP phones registered: 1 Total Registration Statistics Registration requests : 15 Registration success : 15 Registration failed : 0 unRegister requests : 14 unRegister success : 14 unRegister failed : 0

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 timeout interdigit 10 network-locale[0] US (This is the default network locale for this box) network-locale[1] US network-locale[2] US network-locale[3] US network-locale[4] 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] 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

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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 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 username 8701 password cisco6778 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

Pool Tag 2 Config: Mac address is 0019.D111.D2E8 Number list 1 : DN 2 Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled username 8778 password cisco6778 service-control mechanism is not supported registration Call ID is 789d1de7-c8fa-1810-90c4-0019d111d2e8@mat-dtop01tra active primary line is: 8778 contact IP address: 10.67.78.8 port 5061

Dialpeers created: dial-peer voice 40001 voip destination-pattern 8778 session target ipv4:10.67.78.8:5061 session protocol sipv2 codec g711ulaw bytes 160 after-hours-exempt FALSE

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Statistics: Active registrations

: 1

Total SIP phones registered: 1 Total Registration Statistics Registration requests : 15 Registration success : 15 Registration failed : 0 unRegister unRegister requests success unRegister failed

: : 14 14 : 0

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 Active registrations : 1 Total SIP phones registered: 1 Total Registration Statistics Registration requests : 3098 Registration success : 16 Registration failed : 3082 unRegister requests : 15 unRegister success : 15 unRegister failed : 0 Register pool 1 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 Register pool 2 statistics Active registrations : 1 Total SIP phones registered: 1 Total Registration Statistics Registration requests : 15 Registration success : 15 Registration failed : 0 unRegister requests : 14 unRegister success : 14 unRegister failed : 0

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 inclu ding another active SIP call to number 19252302203. uc01tra#show sip-ua calls SIP UAC CALL INFO

Number of SIP User Agent Client(UAC) calls: 0 SIP UAS CALL INFO Call 1 SIP Call ID State of the call Substate of the call

RouteHub Group, LLC


: [email protected] : SIP_STATE_OPTIONS_WAIT (27) : SUBSTATE_NONE (0)

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Calling Number : Called Number : Bit Flags : CC Call ID : Source IP Address (Sig ): Destn SIP Req Addr:Port : Destn SIP Resp Addr:Port: Destination Name : Number of Media Streams :

asterisk 19252302203 0x40000C 0x104 0x0 63829 6.7.7.73 0.0.0.0:0 10.10.10.146:5060 10.10.10.146 1

Number ofObject Active Streams: RTP Fork : 0 0x0 Media Mode : flow-through Media Stream 1 State of the stream : STREAM_IDLE Stream Call ID : -1 Stream Type : voice+dtmf (1) Negotiated Codec : No Codec (0 bytes) Codec Payload Type : 255 (None) Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 Media Source IP Addr:Port: 6.7.7.73:0 Media Dest IP Addr:Port : 0.0.0.0:0 Orig Media Dest IP Addr:Port : 0.0.0.0:0

Options-Ping

ENABLED:NO

ACTIVE:NO

Call 2 SIP Call ID : 73d743e7-c8fa-1810-90d0-0019d111d2e8@mat-dtop01tra State of the call : STATE_ACTIVE (7) Substate of the call : SUBSTATE_NONE (0) Calling Number : 8778 Called Number : 7700 Bit Flags : 0xC0401C 0x100 0x4 CC Call ID : 63823 Source IP Address (Sig ): 10.67.78.1 Destn SIP Req Addr:Port : 10.67.78.8:5061 Destn SIP Resp Addr:Port: 10.67.78.8:5063 Destination Name : 10.67.78.8 Number of Media Streams : 2 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 : 63823 Stream Type : voice-only (0) Negotiated Codec : g711ulaw (160 bytes) Codec Payload Type : 0 Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 Media Source IP Addr:Port: 10.67.78.1:17952 Media Dest IP Addr:Port : 10.67.78.8:5000 Orig Media Dest IP Addr:Port : 0.0.0.0:0 Media Stream 2 State of the stream : STREAM_DEAD Stream Call ID : -1 Stream Type : voice+dtmf (1) Negotiated Codec : No Codec (0 bytes) Codec Payload Type : 255 (None) Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 Media Source IP Addr:Port: 10.67.78.1:0 Media Dest IP Addr:Port : 0.0.0.0:0 Orig Media Dest IP Addr:Port : 0.0.0.0:0

Options-Ping ENABLED:NO ACTIVE:NO Number of SIP User Agent Server(UAS) calls: 2

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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 ============== ========= === ==================== 0/0/0 - 0/0/1 - 0/0/2 - 0/0/3 - 0/1/0 g711ulaw n S_CONNECT 0/1/1 - 0/1/2 - 0/1/3 - 0/4/0 - 0/4/1 - 50/0/10 .1 g711ulaw n S_CONNECT 50/0/10 .2 - 50/0/20 .1 - 50/0/21 .1 - 50/0/11 .1 - 50/0/12 .1 - 50/0/13 .1 - 50/0/13 .2 - 50/0/1 .1 - 50/0/2 .1 - 50/0/22 .1 - 50/0/22 .2 - 50/0/23 .1 - 50/0/23 .2 - 50/0/24 .2 .1 50/0/24 50/0/25 .1 50/0/25 .2

-

-

-

VPM STATE ====================== FXSLS_ONHOOK FXSLS_ONHOOK FXSLS_ONHOOK FXSLS_ONHOOK FXOLS_OFFHOOK FXOLS_ONHOOK FXOLS_ONHOOK FXOLS_ONHOOK EM_ONHOOK EM_ONHOOK EFXS_CONNECT EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK EFXS_ONHOOK 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 0xF959 35ED 0x863831C0 50/0/10.0 0xF95A 35ED 0x8584E568 0/1/0 1 active call found

DSP/Ch 0/1:1

Called # Codec Dial-peers 2091117170 g711ulaw 20001/100 *2091117170 g711ulaw 100/20001

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 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 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0 IP:10.67.78.36 Telecaster 7960 keepalive 1676 1:10 2:12

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ephone-6 Mac:0011.932B.8B15 TCP socket:[2] activeLine:1 REGISTERED mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 debug:0 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 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0 IP:10.67.78.111 7920 keepalive 53081 1:10 2:12 Max 14, Registered process 3, Unregistered ephone_send_packet switched0, 0 Deceased 1, Sockets 4

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 username username username username username username username username username

RouteHub RouteHub RouteHub RouteHub RouteHub RouteHub RouteHub RouteHub RouteHub RouteHub

profile profile profile profile profile profile profile profile profile profile

RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile RouteHub-vm-profile

email email email email email email email email email email

address [email protected] enable preference all attach schedule day 1 active from schedule day 2 active from schedule day 3 active from schedule day 4 active from schedule day 5 active from schedule day 6 active from

01:00 01:00 01:00 01:00 01:00 01:00

to to to to to to

24:00 24:00 24:00 24:00 24:00 24:00

username RouteHub profile RouteHub-vm-profile email schedule day 7 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: Notification Preference: Connection Timeout: Login to VoiceMail allowed: Attach voice message:

enabled All 48 seconds no 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 Message Notification: enabled 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

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Message Notification: enabled Profile: RouteHub-vm-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

[email protected]

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 Profile: RouteHub-vm-profile Device: Email Inbox Enabled: Yes Preference: All Email address: [email protected] 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 "dn6700" 1 0 1 0 "mthomati" 1 1 0 0 "RouteHub" 0 0 0 0 "ucm6_mthomati" 0 0 0 0 "ucm6_RouteHub" 0 0 0 0 "ucm6_sip1" 0 0 0 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: general delivery mailboxes: orphaned mailboxes: capacity of voicemail (minutes): allocated capacity (minutes): total message time used (seconds): total message count: average message length (seconds): broadcast message count: future message count: networking message count: greeting time used (seconds): greeting count: average greeting length (seconds): total time used (seconds):

RouteHub Group, LLC


0 0 0 0 0 0

FUTR

MSGTIME

MBXSIZE

USED

0 0 0 0 0 0

17 25 20 0 0 0

420 420 500 100 100 100

4 6 4 0 0 0

% % % % % %

6 0 0 840 27.333333333333332 35 3 11.666666666666666 0 0 0 25 2 12.5 60

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total time used (minutes): percentage used time (%): messages left since boot: messages played since boot: messages deleted since boot:

1.0 1 27 34 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. 2. 3. 4. 5. 6. 7.

User Error Software Error or Failure Hardware Error or Failure Power Error or Failure Traffic Increase Security Related 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.

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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.

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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 RouteHub Group, LLC


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subject-name cn=IOS-Self-Signed-Certificate-781422512 revocation-check none rsakeypair TP-self-signed-781422512 ! crypto pki trustpoint Equifax_Secure_CA revocation-check none ! crypto pki trustpoint NetworkSolutions_CA revocation-check none ! crypto pki trustpoint trps1_server revocation-check none ! ! crypto pki certificate chain TP-self-signed-781422512 certificate self-signed 01 3082024A 308201B3 A0030201 02020101 300D0609 2A864886 30312E30 2C060355 04031325 494F532D 53656C66 2D536967 69666963 6174652D 37383134 32323531 32301E17 0D303930 315A170D 32303031 30313030 30303030 5A303031 2E302C06 532D5365 6C662D53 69676E65 642D4365 72746966 69636174 35313230 819F300D 06092A86 4886F70D 01010105 0003818D D207FB3A 5F28207E 0A83D6F6 CEACF323 AD0AC7B2 920D1815 9FA0A4EB E0EAA50A 21D70A8F B1E222C6 F06C28B8 5BA19DD3 51AA8D7A 890C4CD3 CA223CEE 02030100 01A37430 72300F06 11041830 16821475 63303174 1D230418 30168014 3D86C4EF 551D0E04 1604143D 86C4EFA3 864886F7 0D010104 05000381 F1F811EB F6609B96 EC536B99 F4B066FB 6FB2D7A6 8CC29B6A 903CD8B7 5B5B21D6 C54E6F37 A2E1D99A 7B7EFDB4 C090BEBD quit crypto pki certificate chain certificate ca 35DEF4CF

855C3BAB ACCCBA10 48A002EF 03551D13 72612E72 A3175A13 175A1331 810002A9 1E5C8CAD 8D9C667D F6D922E3 8333

7AAB90EC D053D828 81C31A14 0101FF04 6F757465 313A16B9 3A16B9A4 EE2F5747 B02B4B3C 8F56C49A EA005478

F70D0101 6E65642D 37313030 03550403 652D3738 00308189 A304DAC4

04050030 43657274 39333334 1325494F 31343232 02818100 9372BA55

80A13600 B07EBF92 3DB01926 05300301 6875622E A482C5B5 82C5B5A1 3784FB27 FCCAD113 3A906851 8154A24A

4310B8C0 ABDF86B7 D1F4A7CA 01FF301F 636F6D30 A19A1E74 9A1E7430 9A34D23D D9B41DDB C3B6E636 82393601

9377678D 2B119217 F31763DB 0603551D 1F060355 301D0603 0D06092A 099369BB 1526317A 7E5E5FDF 164C91FF

Equifax_Secure_CA

30820320 0500304E 66617831 74696669 35315A17 53311030 75696661 74793081 5DB15867 05E0B801 E100144F 4D5930AC 03010001 5D310B30 312D302B

30820289 310B3009 2D302B06 63617465 0D313830 0E060355 78205365 9F300D06 0862EEA0 F04E34EC FCFBF00C 511E3BAF A3820109 09060355 06035504

A0030201 06035504 0355040B 20417574 38323231 040A1307 63757265 092A8648 9A2D1F08 E28A9504 DD43BA5B 2BD6EE63 30820105 04061302 0B132445

02020435 06130255 13244571 686F7269 36343135 45717569 20436572 86F70D01 6D911468 64ACF16B 2BE11F80 457BC5D9 30700603 55533110 71756966

DEF4CF30 53311030 75696661 7479301E 315A304E 66617831 74696669 01010500 980A1EFE 535F05B3 70991557 5F50D2E3 551D1F04 300E0603 61782053

0D06092A 0E060355 78205365 170D3938 310B3009 2D302B06 63617465 03818D00 DA046F13 CB6780BF 9316F10F 500F3A88 69306730 55040A13 65637572

864886F7 040A1307 63757265 30383232 06035504 0355040B 20417574 30818902 846221C3 42028EFE 976AB7C2 E7BF14FD 65A063A0 07457175 65204365

0D010105 45717569 20436572 31363431 06130255 13244571 686F7269 818100C1 D17CCE9F DD0109EC 68231CCC E0C7B902 61A45F30 69666178 72746966

69636174 0603551D 0F040403 20104F33 104F3398 4100040D 03818100 B692639E 2AA72349 D6FA2A66 AC077738

65204175 10041330 02010630 98909FD4 909FD430 300B1B05 58CE29EA 5095D19A 01048642 27A00DFA

74686F72 11810F32 1F060355 301D0603 0C060355 56332E30 FCF7DEB5 6FE411DE 7BFCEE7F A7735CEA

69747931 30313830 1D230418 551D0E04 1D130405 63030206 CE02B917 63856E98 A21652B5 70F19421

0D300B06 38323231 30168014 16041448 30030101 C0300D06 B585D1B9 EEA8FF5A 6767D340 65445FFA

03550403 36343135 48E668F9 E668F92B FF301A06 092A8648 E3E095CC C8D355B2 DB3B2658 FCEF2968

13044352 315A300B 2BD2B295 D2B295D7 092A8648 86F70D01 25310D00 667157DE B228773D A9A28779

4C31301A 0603551D D747D823 47D82320 86F67D07 01050500 A6926E7F C021EB3D AE147761 EF79EF4F

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quit crypto pki certificate chain NetworkSolutions_CA certificate ca 10E776E8A65A6E377E050306D43C25EA 308204A6 3082038E A0030201 02021010 E776E8A6 5A6E377E 0D06092A 864886F7 0D010105 05003081 97310B30 09060355 30090603 55040813 02555431 17301506 03550407 130E5361 43697479 311E301C 06035504 0A131554 68652055 53455254 776F726B 3121301F 06035504 0B131868 7474703A 2F2F7777

050306D4 04061302 6C74204C 52555354 772E7573

3C25EA30 5553310B 616B6520 204E6574 65727472

7573742E 48617264 30313034 13184E65 55040313 74652041 82010F00 167FF19F 23C67E6C F0CE002E 2E3A606F 7B6EEF0C E48542F2 0A2806B7 BAC40930

636F6D31 77617265 3833385A 74776F72 274E6574 7574686F 3082010A 29F6FD03 CCBCA1E9 34A5C8E6 0CA6D9B3 87FB5064 8950E13A CB314125 A6E17502

1F301D06 301E170D 3062310B 6B20536F 776F726B 72697479 02820101 F1ED4D26 7C5046E0 2F0FEC0D F62A2E03 E84E4BEF BE15E345 618B01E9 51B95E64

03550403 30363034 30090603 6C757469 20536F6C 30820122 00C3DD36 9A56F0B5 BD14AD65 EA446175 12D52642 E7719B83 25E25ACB 56A2F63E 8B020301

13165554 31303030 55040613 6F6E7320 7574696F 300D0609 CC83C318 1A1ACDE6 12C20B11 68E5E4DC 0751B264 6361C932 8C3FE033 5F2FF3C4 0001A382

4E2D5553 30303030 02555331 4C2E4C2E 6E732043 2A864886 55B096D9 CC855540 69520A07 80364FDA 5771DC21 8D8CEC14 1E35095A 43F61994 01203082

45524669 5A170D32 21301F06 432E3130 65727469 F70D0101 1325D326 A4B5D00D 921F736F 785D5325 1C89C769 A7E489AD 84EA7E5D 75834CA1 011C301F

7273742D 30303533 0355040A 302E0603 66696361 01050003 864838BB CA22EF3D C1BAD762 9494F54F A3E6FBC2 3F2B2664 A1F59180 82423AC6 0603551D

23041830 1D0E0416 0F0101FF 0603551D 043D303B 2E636F6D 5506082B 703A2F2F 4E416464 01010505 FF0F49AC 790E7EA2 A9EF324A 920F5CF4 18D32B12

168014A1 04143C41 04040302 20041230 3039A037 2F55544E 06010505 7777772E 54727573 00038201 FFE49FD7 5E34184F F0839F73 FE17F195 5B1D281D

725F261B E28F0808 01063012 10300E06 A0358633 2D555345 07010104 75736572 74536572 010068AB 417CA3C5 DF54F1BD 910CA43E 0847522C 7871F6CD

28984395 A94C2589 0603551D 0C2B0601 68747470 52466972 49304730 74727573 7665725F FCEF806B A2E8AAE0 687CE3D3 2B3151A6 508FE89B 36A2E907

5D0737D5 8D6DC538 130101FF 0401860E 3A2F2F63 73742D48 4506082B 742E636F 43412E63 18B2B0B3 57212DC3 D7465E6D 628F1584 A5EEAE70 48443BE7

85969D4B D0FC858C 04083006 01020103 726C2E75 61726477 06010505 6D2F6361 7274300D A34589CB AA7C0C4C 64C2F76D F9A63A12 33899182 576E820A

D2C34530 6217300E 0101FF02 01304406 73657274 6172652E 07300286 63657274 06092A86 53C5A2E6 280B79F4 8882730C 303FDA6E FE30AA76 ADC58ADD

1D060355 0603551D 01003019 03551D1F 72757374 63726C30 39687474 732F5554 4886F70D AF08A9FD EE4C32AD EF9985EA F8CCC719 7659D76C E853B471

AF13D206 9D376D53 3F8A3508 9505FB0A 44C99FA9 40254B32 6F645ED3 B7902E8B 21D8 quit crypto pki certificate chain certificate ca 00 3082029F 30820208 02010030 09060355 04061302 55533111 0E060355 04071307 426F756C 53797374 656D7331 0C300A06 74727073 312D626C 64722E63 01090116 0F777473 75694063 32363231 5A170D30 39303731 13025553 3111300F 06035504 1307426F 756C6465 72311630 73310C30 626C6472 74737569 03818D00 7C099395 09535A4B CBC072B2 DA09FBB3 818100AC 8F9469D2 8E632BBA

0A060355 2E636973 40636973 30818902 997043C9 11EB4BE8 A9B537C0 6E67D8BF C6185869 B344641D C734D76A

RouteHub Group, LLC


040B1303 636F2E63 636F2E63 818100BF B9C4BCF6 B46187CB 84C9F873 F6811102 1324F6BD 75E4A566 266E6A45

FAFEA216 E6B96F5C 5639D6C6 AAEF1967 CE13C5B8 11AF07FE 08D54271 E9E1538B 151FDD2A 07957024

trps1_server 0D06092A 300F0603 64657231 0355040B 6973636F 6973636F 39323032 08130843 14060355

864886F7 55040813 16301406 13035354 2E636F6D 2E636F6D 3632315A 6F6C6F72 040A130D

0D010104 08436F6C 0355040A 47311D30 311E301C 301E170D 30819731 61646F31 43697363

05003081 6F726164 130D4369 1B060355 06092A86 30363130 0B300906 10300E06 6F205379

97310B30 6F311030 73636F20 04031314 4886F70D 32333230 03550406 03550407 7374656D

53544731 6F6D311E 6F6D3081 F80B7E13 DF97F091 BBD9FECB 8A141ED9 03010001 728A8D00 BCB06ACE 88DC366F

1D301B06 301C0609 9F300D06 19C5AA37 0ECB7D06 CB03AE65 D8D15186 300D0609 CEDF15E3 21DFC2B3 C5E12E9E

03550403 2A864886 092A8648 D7433EDC F1B336C6 8F2C5E7E F7047400 2A864886 14671016 041A961C 087AC3AA

13147472 F70D0109 86F70D01 4EC5CAD8 CD134A67 40A66FF2 BB8A2CA1 F70D0101 90ED8F7B 8A23610A 7FEE2089

7073312D 01160F77 01010500 40BEE950 826B0182 899E2FF1 C59DEAD8 04050003 5FF72860 284BC399 C97821A7

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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 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 ! ! no ip bootp server ip domain lookup source-interface BVI10 ip domain name routehub.com ip name-server 206.13.28.12 ip name-server 64.169.140.6 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 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 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 ! RouteHub Group, LLC


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! ! ! ! ! ! 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 ! ! ! ! ! ! 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 ! voice register dn 1 number 8701 name Routehub SIP client (X-lite) ! voice register dn 2 number 8778 name Michel Thomatis (SIP) ! voice register pool 1 id mac 000C.F179.1682 number 1 dn 1 username 8701 password cisco6778 codec g711ulaw ! voice register pool 2 id mac 0019.D111.D2E8 number 1 dn 2 username 8778 password cisco6778 codec g711ulaw ! voice logout-profile 1 pin 6778 user 16778 password 6778 number 6700,A5001,7700,1001,2001 type feature-ring ! voice logout-profile 2 pin 6700 user 16700 password 6700 number 6700,A5002,1002,2002 type feature-ring ! voice user-profile 1 pin 6778 RouteHub Group, LLC


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user 78 password 78 number 6700,A5001,7700,1001,2001 type feature-ring ! voice user-profile 2 pin 6700 user 70 password 70 number 6700,A5002,1002,2002 type feature-ring ! ! voice rule rule ! voice rule ! voice rule ! voice rule rule rule ! voice rule ! voice rule ! voice rule ! voice rule ! ! voice

translation-rule 2 1 /^8(.*)/ /\1/ 2 /^8$1[2-9].........$$/ /\1/ translation-rule 3 1 /^.*/ /19252302203/ translation-rule 4 1 /^4$91[2-9].........$$/ /\1/ translation-rule 5 1 reject /8002197425/ 2 reject /2402107113/ 3 reject /4345339030/ translation-rule 10 1 /^.*$......$/ /\1/ translation-rule 13 1 /19252302203/ /7700/ translation-rule 191 1 /^.*$....$/ /\1/ translation-rule 192 1 /^.*$....$/ /\1/

translation-profile call_block

translate calling 5 ! voice translation-profile translate called 192 ! voice translation-profile translate called 4 ! voice translation-profile translate calling 3 translate called 2 ! voice translation-profile translate called 13 !

routehub-tp-ucm7-outgoing

routehub-tp-s3-external

routehub-tp-sip-outgoing

routehub-tp-sip-outgoing2

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 ! RouteHub Group, LLC


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fax mta mta mta mta mta mta mta

interface-type fax-mail send server 10.67.78.6 port 25 send subject You Received a Fax! send with-subject both send postmaster [email protected] send mail-from hostname routehub.com send mail-from username IncomingFax 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 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 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 queue-manager-debugs 1 param aa-hunt2 6721 param number-of-hunt-grps 1 ! ! ! ! ! spanning-tree backbonefast spanning-tree vlan 1 priority 1000 spanning-tree vlan 10 priority 1000 spanning-tree vlan 11 priority 1000 spanning-tree vlan 100 priority 1000 spanning-tree vlan 101 priority 1000 spanning-tree vlan 102 priority 1000 spanning-tree vlan 999 priority 1000 vtp domain ROUTEHUB vtp mode transparent username mthomati privilege 15 secret 5 $1$.mR8$tKhJRqb8V0JpofbXKIlpg0 ! ! ! RouteHub Group, LLC


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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 ! interface FastEthernet0/0 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 ip virtual-reassembly load-interval 30 speed 100 full-duplex snmp trap ip verify drop-rate no cdp enable crypto map ezvpn RouteHub Group, LLC


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! interface Integrated-Service-Engine0/0 description RHG: CUE interface ip unnumbered BVI10 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 ! interface FastEthernet0/1/0 switchport access vlan 10 macro description cisco-phone auto qos voip trust spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/1 switchport access vlan 10 load-interval 30 macro description cisco-desktop spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/2 switchport access vlan 10 load-interval 30 duplex full auto qos voip trust spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/3 switchport access vlan 10 load-interval 30 macro description cisco-desktop spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/4 switchport access vlan 10 load-interval 30 macro description cisco-desktop spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/5 switchport access vlan 10 load-interval 30 macro description cisco-desktop spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/6 switchport access vlan 101 load-interval 30 macro description cisco-desktop auto qos voip trust spanning-tree portfast service-policy output AutoQoS-Policy-Trust ! interface FastEthernet0/1/7 switchport trunk native vlan 10 switchport trunk allowed vlan 1,2,10,11,1002-1005 switchport mode trunk RouteHub Group, LLC


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! interface FastEthernet0/1/8 switchport mode trunk 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 spanning-disabled 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 bridge-group 10 spanning-disabled ! 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 RouteHub Group, LLC


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ip virtual-reassembly ! ! 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 deny permit permit permit permit permit permit permit permit permit permit permit permit permit

ip 172.16.0.0 0.15.255.255 any ip 192.168.0.0 0.0.255.255 any icmp any any echo-reply icmp any any time-exceeded udp any eq domain any tcp any host 169.1.1.77 eq 3389 8080 www smtp 81 22 tcp any host 169.1.1.73 eq 1723 telnet 22 4662 10000 tcp any host 169.1.1.77 eq 5800 5900 tcp any host 169.1.1.77 eq 9996 udp any host 169.1.1.73 eq non500-isakmp tcp any eq ftp-data host 169.1.1.73 tcp any host 169.1.1.75 eq ftp udp any eq ntp any udp any host 169.1.1.73 eq isakmp 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 ! RouteHub Group, LLC


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logging trap debugging logging 10.67.78.8 access-list 10 permit 192.168.1.0 0.0.0.255 access-list 10 permit 192.168.2.0 0.0.0.255 access-list 10 permit 192.168.3.0 0.0.0.255 access-list 10 permit 10.1.1.0 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:term11.default.loads tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server

flash:S00104000100.sbn flash:cmterm_7936.3-3-5-0.bin flash:P0030702T023.loads flash:P0030702T023.sbn flash:P0030702T023.sb2 flash:P0030702T023.bin flash:cmterm-7941-7961-sccp.7.0.3.tar flash:cnu41.2-7-6-26.sbn flash:CVM41.2-0-2-26.sbn flash:Jar41.2-9-2-26.sbn flash:TERM41.7-0-3-0S.loads flash:term41.default.loads flash:term61.default.loads flash:cnu70.2-7-6-26.sbn flash:CVM70.2-0-2-26.sbn

tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server

flash:Jar70.2-9-2-26.sbn flash:TERM70.7-0-3-0S.loads flash:term70.default.loads flash:term71.default.loads flash:apps31.8-1-0-89.sbn flash:cnu31.8-1-0-89.sbn flash:cvm31sccp.8-1-0-90.sbn flash:dsp31.8-1-0-89.sbn flash:jar31sccp.8-1-0-90.sbn flash:SCCP31.8-1-1SR2S.loads flash:term31.default.loads flash:APPS-1.0.1.SBN flash:CP7921G-1.0.1.LOADS flash:GUI-1.0.1.SBN flash:SYS-1.0.1.SBN

tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server

flash:Desktops/320x212x12/List.xml flash:apps70.8-3-3-17.sbn flash:cnu70.8-3-3-17.sbn flash:cvm70sccp.8-3-3-17.sbn flash:dsp70.8-3-3-17.sbn flash:jar70sccp.8-3-3-17.sbn flash:SCCP70.8-3-4SR1S.loads flash:RingList.xml flash:DistinctiveRingList.xml flash:24.raw flash:24ad.raw

RouteHub Group, LLC


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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 protocol ieee bridge 10 route ip bridge 11 protocol ieee bridge 11 route ip bridge 100 protocol ieee bridge 100 route ip bridge 101 protocol ieee bridge 101 route ip bridge 102 protocol ieee bridge 102 route ip bridge 255 protocol ieee 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 supervisory disconnect dualtone pre-connect pre-dial-delay 0 no vad timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 6700 caller-id enable ! voice-port 0/1/1 connection plar opx 6730 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 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 RouteHub Group, LLC


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switchback method graceful ! 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 ! ! ! dial-peer voice 100 pots call-block translation-profile incoming call_block call-block disconnect-cause incoming call-reject destination-pattern 9.T incoming called-number 6700 direct-inward-dial port 0/1/0 ! dial-peer voice 600 voip destination-pattern 6... session protocol sipv2 session target ipv4:10.67.78.2 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 11 voip description **Outgoing Call to SIP Trunk** translation-profile outgoing routehub-tp-sip-outgoing destination-pattern 8[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 ! dial-peer voice 12 voip 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 ! RouteHub Group, LLC


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dial-peer voice 13 voip description **Emergency Outgoing Call to SIP Trunk** translation-profile outgoing routehub-tp-sip-outgoing destination-pattern 8911 voice-class codec 1 ! dial-peer voice 971 voip translation-profile outgoing routehub-tp-ucm7-outgoing destination-pattern 46... session protocol sipv2 session target ipv4:10.67.78.97 session transport tcp dtmf-relay sip-kpml codec g711ulaw ! dial-peer voice 14 pots service stcapp port 0/0/0 ! dial-peer voice 9011 pots description RP-9011! INTERNATIONAL destination-pattern 9.[0][1][1]T port 0/1/0 ! dial-peer voice 3000 voip description ** UNIFIED MESSAGING ** destination-pattern 671. session protocol sipv2 session target ipv4:10.67.78.92 session transport tcp dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 123 voip incoming called-number 800[0,1]..... codec g711ulaw no vad ! dial-peer voice 1009 voip service aa destination-pattern 6720 session target ipv4:10.67.78.1 incoming called-number 6720 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 7 mmoip description FAX service fax_on_vfc_onramp_app out-bound destination-pattern 6700 information-type fax session target mailto:[email protected] ! dial-peer voice 101 pots service onramp incoming called-number 6730 direct-inward-dial port 0/1/1 ! dial-peer voice 972 voip translation-profile outgoing routehub-tp-ucm7-outgoing destination-pattern 3.... session protocol sipv2 RouteHub Group, LLC


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session target ipv4:10.67.78.191 session transport tcp 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 ip source-address 10.67.78.1 port 2000 auto assign 10 to 19 calling-number initiator timeouts interdigit 5 system message RouteHub UC520 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 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 7961GE TERM41.7-0-3-0S load 7970 term70.default load 7971 TERM70.7-0-3-0S time-zone 5 live-record 6005 voicemail 6000 max-conferences 8 gain -6 call-forward pattern .T call-forward system redirecting-expanded hunt-group report url suffix 0 to 200 hunt-group report every 2 hours moh music-on-hold.au web admin system name admin secret 5 $1$ucO7$XfKgADX7L1nzz11jbTDa./ dn-webedit time-webedit transfer-system full-consult dss transfer-pattern 9.T secondary-dialtone 9 RouteHub Group, LLC


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directory entry 1 919252302203 name ROUTEHUB GROUP (Main) directory entry 2 912098329950 name Deliver Ease (Main) fac standard create cnf-files version-stamp 7960 Mar 10 2009 14:54:25 ! ! 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 ! ! ephone-dn 3 dual-line ring internal number 1001 no-reg primary label 1001 (Office) name 1001 call-forward busy 6710 call-forward noan 6710 timeout 15 ! ! ephone-dn 4 dual-line ring internal number 1002 no-reg primary label 1002 (Family Room) name 1002 mobility snr 812096277170 delay 2 timeout 30 cfwd-noan 6000 call-forward noan 6700 timeout 30 ! ! ephone-dn 5 number 3001 call-forward all 4001 ! ! ephone-dn 10 dual-line ring external number 6700 no-reg primary label 6700 (Main) description 2098322559 name 6700 call-forward busy 6000 call-forward noan 6000 timeout 15 ! ! ephone-dn 11 number A5001 no-reg primary label Intercom name Intercom intercom A5002 RouteHub Group, LLC


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! ! ephone-dn 12 number A5002 no-reg primary label Intercom name Intercom intercom A5001 ! ! ephone-dn 13 dual-line ring external number 7700 secondary 19252302203 label 7700 (RouteHub Main) name 7700 call-forward busy 6000 call-forward noan 6000 timeout 15 ! ! ephone-dn 14 dual-line number 2001 label 2001 (Agent) ! ! ephone-dn 15 dual-line number 2002 label 2002 (Agent) ! ! ephone-dn 16 number 6005 call-forward all 6000 ! ! ephone-dn 20 number 8000.... no-reg primary mwi on ! ! ephone-dn 21 number 8001.... no-reg primary mwi off ! ! ephone-dn 22 dual-line number 6999 conference meetme no huntstop ! ! ephone-dn 23 dual-line number 6999 conference meetme preference 1 no huntstop ! ! ephone-dn 24 dual-line number 6999 conference meetme preference 2 no huntstop ! ! ephone-dn 25 dual-line RouteHub Group, LLC


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number 6999 conference meetme preference 3 no huntstop ! ! ephone-dn 26 dual-line number 6998 name Conference conference ad-hoc preference 1 no huntstop ! ! ephone-dn 27 dual-line number 6998 name Conference conference ad-hoc preference 2 no huntstop ! ! ephone 1 device-security-mode none mac-address 001B.D52C.77C5 type 7906 button 1:13 ! ! ! ephone 2 device-security-mode none video mac-address 001C.58F0.7619 paging-dn 1 type 7970 logout-profile 2 ! ! ! ephone 3 device-security-mode none mac-address 000C.299C.8EE7 type CIPC ! ! ! ephone 4 device-security-mode none mac-address D456.7C69.0000 max-calls-per-button 2 type anl button 1:10 ! ! ! ephone 5 device-security-mode none mac-address 0012.00A7.72EA username "dn6700" paging-dn 1 type 7960 button 1:10 2:12 ! RouteHub Group, LLC


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! ! ephone 6 device-security-mode none 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 device-security-mode none 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... || || || || |||| |||| ..:||||||:..:||||||:.. c i s c o S y s t e m s OFFICAL USE ONLY! RouteHub Group, LLC (925) 230-2203 www.routehub.com [email protected] ----------------------------------------------------------------------^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 preferred none transport input all RouteHub Group, LLC


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line vty 0 4 transport preferred none 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" ip name-server 10.67.78.1 ntp server 69.31.13.210 prefer software download server url "ftp://10.67.78.243/cue7" credentials hidden "c9yem6kD1vLmzOwzb80QtWTRRNhfheJsSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9 RouteHub Group, LLC


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J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP" site name local site-hostname 10.67.78.1 web credentials hidden "IRBD7mOWpxpWaKsZ0zTwvt+jA8JZJIp2Sd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9 J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP" end site fax gateway outbound address 10.67.78.1 privilege privilege privilege privilege privilege privilege privilege privilege privilege privilege

ViewPrivateList create ViewRealTimeReports create manage-passwords create ManagePrompts create broadcast create vm-imap create ManagePublicList create manage-users create ViewHistoricalReports create local-broadcast create

groupname Broadcasters create username username username username username username username username username username

ucm6_routehub create ucm6_sip1 create administrator create dn6700 create admin create cisco create routehub create Fax create ucm6_mthomati create 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 privilege privilege privilege privilege privilege privilege privilege privilege privilege privilege

manage-passwords operation user.pin manage-passwords operation user.password manage-passwords operation system.debug ManagePrompts operation system.debug ManagePrompts operation prompt.modify broadcast operation system.debug broadcast operation broadcast.remote broadcast operation broadcast.local vm-imap operation voicemail.imap.user ManagePublicList operation voicemail.lists.public ManagePublicList operation system.debug

RouteHub Group, LLC


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privilege privilege privilege privilege privilege privilege privilege privilege

manage-users manage-users manage-users manage-users manage-users manage-users manage-users manage-users

operation operation operation operation operation operation operation operation

user.pin group.configuration user.mailbox user.configuration user.notification user.password system.debug user.remote

privilege ViewHistoricalReports operation report.historical.view privilege local-broadcast operation system.debug privilege local-broadcast operation broadcast.local groupname groupname groupname groupname groupname groupname groupname

Administrators member admin Administrators member routehub Administrators member administrator Administrators member mthomati Administrators member cisco Administrators member dn6700 Broadcasters privilege broadcast

username ucm6_routehub phonenumber "106701" username ucm6_sip1 phonenumber "106801" username dn6700 phonenumber "6700" username username username username username

routehub phonenumber "7700" Fax phonenumber "6730" ucm6_mthomati phonenumber "106778" mthomati phonenumber "6778" routehub phonenumberE164 "19252302203"

restriction restriction restriction restriction

msg-notification msg-notification msg-notification msg-notification

create min-digits 1 max-digits 30 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 J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP" calendar biz-schedule systemschedule open day 1 from 00:00 to 24:00 open day 2 from 00:00 to 24:00 open day 3 from 00:00 to 24:00 open day 4 from 00:00 to 24:00 open day 5 from 00:00 to 24:00 open day 6 from 00:00 to 24:00 open day 7 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" RouteHub Group, LLC


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parameter "PlayExtensionsPrompt" "AASPlayExtensions.wav" parameter "ExtensionLength" "4" end application ccn application ciscomwiapplication aa description "ciscomwiapplication" enabled maxsessions 4 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "8001" parameter "strMWI_ON_DN" "8000" end application ccn application msgnotification aa description "msgnotification" enabled maxsessions 6 script "msgnotify.aef" parameter "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp" 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 "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp" 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 gateway address "10.67.78.1" 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 RouteHub Group, LLC


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ccn trigger sip phonenumber 6000 application "voicemail" enabled maxsessions 6 end trigger ccn trigger sip phonenumber 6003 application "autoattendant" enabled locale "en_US" maxsessions 6 end trigger ccn trigger sip phonenumber 6006 application "promptmgmt" enabled idletimeout 5000 locale "en_US" maxsessions 1 end trigger service phone-authentication end phone-authentication service voiceview enable end voiceview voicemail voicemail voicemail voicemail

notification notification notification notification

enable preference all allow-login email attach


callerid configuration outgoing-email from-address [email protected] default language en_US 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 RouteHub Group, LLC


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description "ucm6_mthomati messagesize 60 end mailbox

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


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