voavpn cucme 7.0(1) ccg - at&t › bvoip › wp-content › themes … · web viewip address...

AT&T IP Flexible Reach on AT&T VPN Cisco & Unified Communications Manager Express Customer Configuration Guide

IP Flexible Reach on AT&T VPNCisco Unified Communications Manager Express 7.0(1)

Customer Configuration Guide

Issue 1.03/7/2011

1


Table of Contents1 Introduction..................................................................................................................3

1.1 Overview................................................................................................................31.2 Cisco Unified Communications Manager Express Site.........................................41.3 Features Supported.................................................................................................51.4 Features not supported............................................................................................61.5 Platforms & Access Types.....................................................................................6

2 AT&T VPN WAN Access Configuration...................................................................62.1 T1 PPP....................................................................................................................62.2 NxT1 MLPPP (2-8 T1s).........................................................................................72.3 Traffic Classification and Queuing Techniques...................................................102.4 Class of Service Configuration Example.............................................................122.5 Routing.................................................................................................................13

2.5.1 Additional Instructions for the Voice Quality Monitor.................................132.5.2 Probe IP Addressing & Routing....................................................................142.5.3 TAP................................................................................................................16

3 Cisco Unified Communications Manager Express Configuration............................163.1 Cisco Unified Communications Manager Express Version.................................163.2 DHCP Server........................................................................................................173.3 NTP Server...........................................................................................................183.4 Install CME Application Files..............................................................................183.5 CME Telephony Parameters................................................................................203.6 Dialing Restrictions..............................................................................................233.7 Phone Configuration.............................................................................................25

3.7.1 Ephone-dn......................................................................................................253.7.2 Ephone...........................................................................................................26

3.8 Conference............................................................................................................273.9 VoIP Connectivty to AT&T IP Flexible Reach...................................................313.10 Codec configuration..........................................................................................313.11 SIP Profile.........................................................................................................323.12 Privacy with SIP (Optional)..............................................................................323.13 Calling/Called Party Number Translation Configuration.................................333.14 Outbound Calls to AT&T.................................................................................343.15 Inbound Calls from AT&T...............................................................................36

4 Unity Express Configuration.....................................................................................365 Troubleshooting.........................................................................................................396 Appendix A: Acronyms.............................................................................................64

2


1 IntroductionAT&T IP Flexible Reach on AT&T Virtual Private Network (AT&T VPN) Service is an offering that allows calls from a customer managed VoIP environment to connect to the PSTN and offers the customer a viable alternative to traditional PSTN. This document introduces support for Cisco Call Manager Express (CME) for customer’s who wish to deploy Cisco Integrated Service Router with CME application at their site. The CME software is a call processing application in the Cisco IOS that enables the Cisco router to deliver PBX like functionality along with WAN termination for small branch offices within the customer’s IP Telephony network

This application note describes how to configure Cisco Unified Communications Manager Express (CUCME) 7.0(1) on a Cisco Integrated Service Router (ISR) for connectivity to AT&T’s IP Flexible Reach on AT&T VPN service (Calling plans: IP Long Distance and IP Local).

Laboratory testing was performed for the preparation of this guide. Key features verified were: Inbound/Outbound Basic Calls, Calling Name delivery, Codec Negotiation, Intra-site Transfers, Intra-site Conferencing, Call Hold and Resume, Call Forward (forward all, busy and no answer), conferencing, inbound and outbound calls to TDM networks and Cisco Unfied Communications IP PBX networks, hop-off and hop-on calls to the PSTN.

The CUCME configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper interoperability between AT&T SIP network and Cisco Unified Communications Manager Express. The configuration described in this document details the important commands to have enabled for interoperability to be successful and care must be taken, by the network administrator deploying CUCME, to ensure these commands are set per each dial-peer required to interoperate with the AT&T SIP network.

1.1 Overview

This section provides a service overview of the Cisco Unified Communications Manager Express integration with AT&T IP Flexible Reach. The components are shown next; consult your Cisco representative for the correct Cisco IOS software image for the specific platform and CME application as well as the Feature License requirements for Cisco Unified Communications Manager Express and Cisco Unity Express Elements.

Figure 1 shows the AT&T IP Flexible Reach service topology to support Cisco Unfied Call Manger Express 7.0(1) at remote locations and the HQ site using CUBE and CUCM that was tested for this project.

3


1.2 Cisco Unified Communications Manager Express Site

The Cisco Unified Communications Manager Express (“CUCME”) site consists of the following components.

Hardware Components

Cisco IP Phones (customer managed) – These may be hard phones or soft phones. Although CUCME can also support SIP on the line side to SIP IP Phones, this document does not address such configurations.

Cisco Integrated Service Router (ISR) terminating the AT&T WAN circuit and running the Unified Communication Manager Express (customer managed) IP PBX application with Cisco Unity Express voice mail and auto attendant on an Integrated Service Engine card.

4


Packet Voice Data Module (PVDM). You will need to install DSP modules (PVDM) on the ISR if you require support for analog interfaces, Transcoding or Conference Bridge resources for codecs other than G.711

Software Requirements

Cisco Unified Communications Manager Express 7.0(1) IOS image version 12.4(22)T5. This solution was tested with c2800nm-adventerprisek9-mz.124-22.T5.bin

Cisco Unity Express 7.0(1)

** Special Notes

Emergency 911/E911 Services Limitations and Restrictions - Although AT&T provides 911/E911 calling capabilities, AT&T does not warrant or represent that the equipment and software (e.g., IP PBX) reviewed in this customer configuration guide will properly operate with AT&T IP Flexible Reach to complete 911/E911 calls; therefore, it is Customer’s responsibility to ensure proper operation with its equipment/software vendor.

While AT&T IP Flexible Reach services support E911/911 calling capabilities under certain Calling Plans, there are circumstances when that E911/911 service may not be available, as stated in the Service Guide for AT&T IP Flexible Reach found at http://new.serviceguide.att.com. Such circumstances include, but are not limited to, relocation of the end user’s CPE, use of a non-native or virtual telephone number, failure in the broadband connection, loss of electrical power, and delays that may occur in updating the Customer’s location in the automatic location information database. Please review the AT&T IP Flexible Reach Service Guide in detail to understand the limitations and restrictions.

NOTE: N11 (including 911) calls are not supported unless AT&T IP Flexible Reach Local Service is ordered!!!

1. Please contact your customer care representative for the AT&T IP Border Element Addresses for your specific IP PBX.

1.3 Features Supported

Basic Call using G.729 or G.711ulaw Calling Party Number Presentation

5

http://new.serviceguide.att.com/


Calling Name Intra-site Call Transfer Intra-site Conference Call Hold and Resume Call Forward All, Busy and No Answer Incoming DNIS Translation and Routing Outbound calls to AT&T’s IP and TDM networks

1.4 Features not supported

Fax support on CME

1.5 Platforms & Access Types

Table 1: Cisco ISR Platform IP Telephony and Memory Specifications

Router Platform

Max. IP Phones

Max. Ephone-dnVirtual Ports

Minimum DRAM

Minimum Flash

Cisco 2811 36 144 256 64Cisco 2821 48 144 256 64Cisco 2851 96 288 256 64Cisco 3825 168 500 256 64Cisco 3845 240 720 256 64

Following are the access types that can be ordered:

Access Type Speed (bit/s) Fragmentation CRTPT1 PPP access 1024K to 1536K no no

NXT1 MLPPP access N = 2 to 8 T1s no no

2 AT&T VPN WAN Access Configuration

2.1 T1 PPP

For T1 PPP access configuration, the bandwidth statement must be configured to slightly less than the access speed. The IP address is configured which should be the CER side of the /30 subnet assigned for the CER/PER link. The CoS policy “SHAPE_PPP” is applied to the serial interface. In this case, the max-reserved-bandwidth statement set at 100 will help insure that the service policy is properly applied to the interface.

6


Example of T1 PPP Access on ISR:

interface Serial0/1/0

ip address <ip address> <mask>

max-reserved-bandwidth 100

encapsulation ppp

service-policy output SHAPE_PPP

2.2 NxT1 MLPPP (2-8 T1s)

For N X T1 MLPPP access, each individual T1 interface will need to be configured as part of a multilink group. The first step is define the VWIC2-(1 or 2) MFT-T1/E1 as T1 cards with the command “card type t1 <slot #>”. Then the T1 controllers must be configured as shown below. Once the controller cards are configured, serial interfaces which match the T1 controller numbers will appear. Each of these serial interfaces must be configured to be part of a multilink group/interface. The multilink interface is created first. The IP address and COS policy, “COS_MLPPP”, are applied to the multilink interface. The bandwidth statement on the multilink interface is equal to the derived port speed depending on the number of T1 interfaces. In this case, the max-reserved-bandwidth statement set at 100 will insure that the service policy is properly applied to the interface. PPP CHAP is used with the IP address of the Multilink interface. Then the serial interfaces are configured. The multilink and PPP CHAP commands are applied to each the serial interface into the multilink group.

Example of 4 X T1 MLPPP on ISR:

card type t1 0 0card type t1 0 1!!controller T1 0/0/0 framing esf linecode b8zs channel-group 0 timeslots 1-24!controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24

7


!controller T1 0/1/0 framing esf linecode b8zs channel-group 0 timeslots 1-24!controller T1 0/1/1 framing esf linecode b8zs channel-group 0 timeslots 1-24

interface Multilink1 bandwidth <1536 X # of T1s> ip address <ip address> <mask> load-interval 30 ppp chap hostname <CE IP address> ?? ppp multilink ppp multilink group 1 ppp multilink fragment disable max-reserved-bandwidth 100 service-policy output COS_MLPPP no peer neighbor-route

interface Serial0/0/0:0 no ip address encapsulation ppp load-interval 30 tx-ring-limit 2 tx-queue-limit 2 ppp chap hostname <CE IP address> ppp multilink ppp multilink group 1 !interface Serial0/0/1:0 no ip address encapsulation ppp load-interval 30 tx-ring-limit 2 tx-queue-limit 2 ppp chap hostname <CE IP address> ppp multilink ppp multilink group 1!interface Serial0/1/0:0 no ip address encapsulation ppp tx-ring-limit 2 tx-queue-limit 2 ppp chap hostname <CE IP address> ppp multilink ppp multilink group 1 !interface Serial0/1/1:0 no ip address encapsulation ppp

8


tx-ring-limit 2 tx-queue-limit 2 ppp chap hostname <CE IP address> ppp multilink ppp multilink group 1

interface Loopback0 description "FlexReach assigned Media+Signaling Address" ip address <x.x.x.x 255.255.255.255>!interface Loopback1 description "FlexReach assigned VoIP Probe Address" ip address <x.x.x.x 255.255.255.255>

interface GigabitEthernet0/0 description 802.1Q Trunk to Customer_Switch no ip address duplex full speed 100!interface GigabitEthernet0/0.1 description Branch IP Phone subnet encapsulation dot1Q 70 ip address <x.x.x.x> 255.255.255.0!interface GigabitEthernet0/0.2 description Branch LAN ip address <x.x.x.x> 255.255.255.0 load-interval 30 duplex auto speed auto!interface GigabitEthernet0/0.3 description VoIP Probe Branch LAN ip address 10.255.255.253 255.255.255.252 load-interval 30 duplex auto speed auto ip nat inside!interface Serial0/0/0 description AT&T VPN WAN Access T1 PPP CKT ID ip address <x.x.x.x 255.255.255.252> <CE Address is even PE address is ODD> encapsulation ppp load-interval 30 max-reserved-bandwidth 100 service-policy output SHAPE_PPP ip nat outside!

9


2.3 Traffic Classification and Queuing Techniques

Class of Service features operates in concert with customer router behaviors to provide end-to-end congestion management of application traffic flows. The Customer Premise Equipment (CPE) has several roles in the process. First, it must recognize and categorize the different application types that are to receive differentiated service. Based on this recognition, queuing, fragmentation and interleaving, techniques are used as appropriate to provide preferential treatment of priority traffic during congestion. In addition to the treatment within the CPE, the network needs to recognize and provide differentiated treatment of customer application traffic. To accommodate this, the CPE needs to mark the various application types with appropriate DiffServ codepoints. This allows the network to recognize the different traffic types to provide the desired preferential treatment.

After determining bandwidth requirements and the techniques required to meet the delay budgets, CoS techniques should be applied in the Customer Edge Router(CER) to compliment the functionality in the network PER. CoS techniques will help minimize delay, jitter (variation in delay) and drops of voice packets. These techniques include classifying and marking packets by traffic type, using queuing techniques, and traffic shaping. Classification

The first step in traffic classification is to identify different traffic flows and mark them with the appropriate DSCP bit. The following table defines the settings expected by the AT&T VPN network.

Class of Service IP Precedence DSCP DSCP Decimal

DSCP Binary (In Contract)

Real Time (COS1) 5 EF 46 101 110

Bursty High (COS2) 3 AF31 26 011 010

Bursty Low (COS3) 2 AF21 18 010 010

Best Effort (COS4) 0 BE 0 000 000

Additional Classes for COS6:

Class of Service IP Precedence DSCP DSCP Decimal

DSCP Binary (In Contract)

Video (COS2V) 4 AF41 34 100 010

Scavenger (COS5) 1 AF11 10 001 010

Queuing Options

10


Queuing techniques and implementations have evolved over the past several years and include options that can strictly prioritize voice traffic over data traffic without starving out the data traffic. Strict priority queuing is a mechanism that will always immediately serve any packets in the priority queue before serving any other queue, ensuring the best possible delay characteristics. In the IP Flex on AT&T VPN network, AT&T uses Low Latency Queuing with Class Based Weighted Fair Queuing (LLQ/CBWFQ) and recommends that customers use the same techniques in their edge routers. (Note: An IOS of 12.4.6.T10 is recommended). LLQ/CBWFQ is configured via a policy map where different classes of traffic are assigned a percentage or specific amount of bandwidth. The LLQ is established with the priority command and given a specific bandwidth in kilobits per second. The LLQ is sized based on the bandwidth allocation recommendations in section 2.1. Other queues are serviced based on the amount of bandwidth allocated to them.

Traffic Shaping Typically with ePVCs on low speed frame relay ports, AT&T recommends traffic shaping for voice traffic by setting the router’s CIR slightly less than port speed, depending on how aggressive a strategy is desired. Also, excess bursting should be “turned off” by setting the Burst Excess (Be) parameter to 0. Set the committed burst rate (Bc) such that the time interval (Tc) is close to 10 msec—a good rule of thumb for this is to divide the CIR value by 1000. For example, if shaping to a CIR of 128kbps, set Bc to 1280.

2.4 Class of Service Configuration Example

11


Configuration for class-map and policy-map used while testing CME IOS release 12.4(22)T4 routers.

!class-map match-any COS2V match access-group name COS2V-Trafficclass-map match-any BGP match access-group name BGPclass-map match-any COS5 match access-group name COS5-Trafficclass-map match-any COS4 match access-group name COS4-Trafficclass-map match-any COS3 match access-group name COS3-Trafficclass-map match-any COS2 match access-group name COS2-Traffic match access-group name BGPclass-map match-any COS1 match access-group name RTP match access-group name SIP match access-group name SCCP!policy-map MARK-BGP class BGP set ip dscp cs6policy-map COS class COS1 priority 616 77000 set ip dscp ef class COS2V bandwidth remaining percent 30 set ip dscp af41 class COS2 bandwidth remaining percent 30 set dscp af31 service-policy MARK-BGP class COS3 bandwidth remaining percent 30 set ip dscp af21 class COS5 bandwidth remaining percent 1 set ip dscp af11 class class-default bandwidth remaining percent 9 set ip dscp defaultpolicy-map SHAPE_PPP class class-default shape average 1456000 14560 queue-limit 4096 packets service-policy COS!

!ip access-list extended BGP permit tcp any eq bgp any permit tcp any any eq bgp

12


ip access-list extended COS2-Traffic permit udp any any eq 2082 permit udp any eq 2082 anyip access-list extended COS2V-Traffic permit tcp any any range 3230 3231 permit udp any any range 3230 3235ip access-list extended COS3-Traffic permit udp any any eq 2083 permit udp any eq 2083 anyip access-list extended COS5-Traffic permit udp any any eq 110 permit udp any eq 110 anyip access-list extended RTP permit udp any range 16384 32767 any range 16384 32767ip access-list extended SCCP permit tcp any range 2000 2003 any permit tcp any any range 2000 2003ip access-list extended SIP permit udp any eq 5060 any permit udp any any eq 5060!

2.5 Routing

Configure BGP routing process on the CME router.

!router bgp 65000 no synchronization bgp router-id 192.169.135.1 bgp log-neighbor-changes network 32.x.x.x mask 255.255.255.255 (CME Loopback for FlexReach Signaling) network 32.x.x.x mask 255.255.255.255 (VoIP Probe IP address) network 32.x.x.x mask 255.255.255.255 (CME Loopback IP address used for NAT’ing VoIP probe inside address) neighbor 192.168.135.2 remote-as 13979 (PE Address) neighbor 192.168.135.2 allowas-in (PE Address) no auto-summary!

2.5.1 Additional Instructions for the Voice Quality Monitor

Figure 3 shows the WineyeQ probe topology at the CUCME store location.

13


2.5.2 Probe IP Addressing & Routing

Additional Router Instructions for the Voice Quality Monitor

These additional steps are required for proper routing of the Voice Quality Monitor and must be configured on the Customer Call Manager Express Router (CME).

Step 1) In order to allow for the Voice Quality Monitor to setup the IPSEC tunnel to the security device in the AT&T network (which has an IP address of 32.95.217.109), a static route must be configured as follows on the CME.

Note: The customer should have received the IP address of PER during the initial AT&T VPN provision process.

ip route 32.95.217.109 255.255.255.255 <IP address of PER>

Step 2) Create a new loopback interface, ethernet interface and the necessary network address translation (NAT) for VoIP probe communication.

14


Note: Customer will be given two host addresses for the Voice Quality Monitor configuration. The address with the lower fourth octet will be used for the loopback interface. The address with the higher fourth octet will be used for the static route statement (see step 3) . Example is shown in step 3

int loopback <loopback #>

ip address < IP address assigned by IP Flex for SIP Signaling> 255.255.255.255

int loopback <loopback #>

ip address < IP address assigned by IP Flex with lower fourth octet> 255.255.255.255

int <ethernet interface connecting to the Voice Quality Monitor>

ip address 10.255.255.253 255.255.255.252

ip nat inside

int <use your WAN interface that has IP address associated with it>

ip nat outside

ip nat inside source static 10.255.255.254 < IP address assigned by IP Flex with lower fourth octet>

Step 3) In order to advertise the Voice Quality Monitor’s IP address to the AT&T network, a static route must be added with the Voice Quality Monitor IP address on the CME. Once the customer confirms receipt of the Voice Quality Monitor, the IP address information will be send out to the customer. For the static route, the IP address assigned by IP Flex with the higher fourth octet must be used.

**The static route for the Voice Quality Monitor must have a 32 bit mask**.

ip route <Ip address assigned by IP Flex with higher fourth octet > 255.255.255.255 10.255.255.254

Step 4) Redistibution of Routes, after the setup of the static routes the CME router need to advertise these two usable addresses through BGP

router bgp <your AS number>network 32.252.97.1 mask 255.255.255.255network 32.252.97.2 mask 255.255.255.255

15


2.5.3 TAP

The Tap as seen in the diagram above will not be IP Addressable as this is a pass-thru device connected to the customer’s LAN. The LAN port from the Tap to the WinEyeQ Probe PC will have a default IP addressed specified by Touchstone as this is a read-only port and the IP address assigned only has local significance and will not be propagated anywhere in the network. The connectivity from the WinEyeQ Probe PC to the Customer CPE L2/3 Switch will use a /30 (255.255.255.252) mask allocated from a predefined range allocated by AT&T wherein the first useable address will be defined on the CPE LAN port and the second useable address will be assigned to the Network Interface Card (NIC) of the probe facing the TAP. Finally, the probe will use the assigned address to establish an IPSec tunnel to an aggregator within AT&T’s network for SLA reporting purposes.

**Note: The probe sends its data back to AT&T via a secure L2TP/IPSec tunnel on UDP port 1701. Please ensure this port is not blocked by any access lists on the CME Router.

3 Cisco Unified Communications Manager Express Configuration

This configuration guide specifies the Cisco Unified Communications Manager screens that must be configured and updated to support the IP Flexible Reach on AT&T VPN service.

3.1 Cisco Unified Communications Manager Express Version

Cisco IOS Version

CME1#show versionCisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version 12.4(22)T5, RELEASE SOFTWARE (fc2)Technical Support: http://www.cisco.com/techsupportCopyright (c) 1986-2009 by Cisco Systems, Inc.Compiled Mon 14-Dec-09 19:16 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T11, RELEASE SOFTWARE (fc1)

CME1 uptime is 3 weeks, 6 days, 20 hours, 23 minutesSystem returned to ROM by reload at 11:55:29 EDT Mon Mar 29 2010

16


System restarted at 11:56:55 EDT Mon Mar 29 2010System image file is "flash:c2800nm-adventerprisek9-mz.124-22.T5.bin"

This product contains cryptographic features and is subject to UnitedStates and local country laws governing import, export, transfer anduse. Delivery of Cisco cryptographic products does not implythird-party authority to import, export, distribute or use encryption.Importers, exporters, distributors and users are responsible forcompliance with U.S. and local country laws. By using this product youagree to comply with applicable laws and regulations. If you are unableto comply with U.S. and local laws, return this product immediately.

A summary of U.S. laws governing Cisco cryptographic products may be found at:http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

If you require further assistance please contact us by sending email [email protected].

Cisco 2851 (revision 53.51) with 763904K/22528K bytes of memory.Processor board ID FTX1343AJ6T2 Gigabit Ethernet interfaces2 Serial(sync/async) interfaces1 terminal line1 Virtual Private Network (VPN) Module1 cisco Integrated Service Engine(s) Cisco Unity Express 7.0.5 in slot 1DRAM configuration is 64 bits wide with parity enabled.239K bytes of non-volatile configuration memory.492408K bytes of ATA CompactFlash (Read/Write)

Configuration register is 0x2102

3.2 DHCP Server

Configure the CME router to act as a DHCP server as follows:

Task 1: Enable DHCP server and relay agent on the CME router

service dhcp

Task 2: Configure DHCP address pools. In this example a pool is defined to support the branch store IP Phones. ip dhcp pool Branch-phones !Specify the name of the dhcp pool serving the IP Phones network 10.30.10.0 255.255.255.0 default-router 10.30.10.1 ! Specify the CME VLAN IP address as the default gateway Option 150 ip x.x.x.x ! Specify the CME as the tftp server !

Task 3: Define the ip address range that has to be excluded from the address pool range. In this example, the CME router has 1 VLAN serving the ip phones. Exclude the ip

17


address on those VLANs and other if applicable from the dhcp ip address pool. This will prevent the CME router from reassigning the ip addresses.

Example Configuration

ip dhcp excluded-address 10.30.10.1 10.30.10.75!ip dhcp pool Branch-phones network 10.30.10.0 255.255.255.0 default-router 10.30.10.1 option 150 ip <x.x.x.x> Loopback0 IP Address

3.3 NTP Server

ntp master 2ntp server 10.10.33.1

Network Time Protocol (NTP) is required within the AT&T IP Flexible Reach offer to allow the CME router clock to synchronize to a single master clock source on the network. The phones derive their date and time information from the CME they are registered to. Gather the following information before you configure NTP on the CME router.

Identify the master NTP server ip address on the network Identify the local time zone of the CME site Determine the number of hours that the local time zone differs from the

Coordinated Universal Time (UTC) or Greenwich Mean Time (GMT)

Task 1: Enable NTP and configure the CME to derive its clock source from master NTP server (ip address 10.10.33.1). In this example (i.e. the lab environment) the router is the Master Clock source.

clock timezone EST -5 ! Specify the local timezone (EST) and offset time from UTC in hours (-5) clock summer-time EDT recurring! Enable the daylight saving time.ntp server 10.10.33.1 ! Specify the ip address of the Master clock source on the network.

3.4 Install CME Application Files

The recommended files that you need to install are the following: Basic Files: This is a TAR archive containing the basic files you need to run the

Cisco Unified Call Manager express. This archive contains also the phone firmware files required, although additional individual phone firmware files may be needed sometimes. The filename for this tar archive is “cme-basic-

18


x.x.x.tar“. GUI Files: This is again a TAR archive containing only the files of the GUI

management tool, which is a mouse-driven interface for provisioning phones and for general CME management after basic installation is complete. The filename for this tar archive is “cme-gui-x.x.x.tar“.

Phone Firmware Files: Although the required phone firmware files are included in the Basic tar archive, you may need to add phone firmware files to support individual phone models that are not included in the basic package. Each firmware file is specific for each phone model and for the protocol it uses (i.e SCCP or SIP protocol). By default, new IP phones are shipped with an SCCP firmware image. If the firmware installed on an IP phone is older than the firmware loaded on the Call Manager router flash, the IP phone automatically upgrades its firmware and then registers with the Callmanager. The filename conventions used for phone firmware images are:

SCCP firmware: P003xxyyzzww or SCCPxxyyzzww SIP firmware: P0S3-xx-y-zz or SIPxxyyzzww For Java-based IP phones, such as the Cisco Unified IP Phone 7911,

7941, 7941GE, 7961, 796GE, 7970, and 7971, the firmware consists of multiple files including JAR and tone files

Installation of CME softwareAfter you download the .zip CME software file from Cisco, uncompress the file on a local TFTP server. You will get several individual files and several TAR archives. We assume the TFTP server has access to the CallManager Express router.For individual files:Use the regular copy command to transfer the file from TFTP to the router’s flash:Example:Router# copy tftp://x.x.x.x/P00307020300.sbn flash:For TAR archive files:Use the archive command to transfer the files and extract them at the same time to the router’s flash:Example:To transfer the Basic Files tar archive (cme-basic-3.0.3.tar) to callmanager router:Router# archive tar /xtract tftp://x.x.x.x/7970-backgrounds.tar flash:

After all required files are installed, use the “show flash” command to list the files installed on flash memory.

cme-124-22T2.zip

19


7970-backgrounds.tarapp-cme-did-2.0.0.0.tarcme-bacd-2.1.2.2.tarcmterm-3951-sip.8-1-1.tarcmterm-7911_7906-sccp.8-3-3S.tarCP7921G-1.1.1.TARcmterm-7931-sccp.8-3-3S.tarcmterm-7941_7961-sccp.8-3-3S.tarcmterm-7942_7962-sccp.8-3-3S.tarcmterm-7945_7965-sccp.8-3-3S.tarcmterm-7970_7971-sccp.8-3-3S.tarcmterm-7975-sccp.8-3-3S.tarP00308000500.tarringtone.tarcmterm_7920.4.0-03-02.bincmterm_7936.3-3-15-0.bincmterm_7985.4-1-4-0.binCP7902080002SCCP060817A.sbinCP7912080003SCCP070409A.sbinmusic-on-hold.auP00405000700.binP00405000700.sbnP00503021500.binS00105000300.sbnapps37sccp.1-1-1-1.binATA030203SCCP051201A.zupB015-1-0-2.SBNB016-1-0-2.SBN

3.5 CME Telephony Parameters

Gather the following information before you configure the CME specific parameters that allow the router to communicate with the phones.

Identify the maximum number of phones that the CME router can support. The maximum is dependent on router platform and CME ver. The IPT offer development team recommends that this number to be less that the maximum number a given platform can support.

Work with the customer to define the system message that will be displayed on all phones.

Identify the firmware load files for each phone in the customer’s IPT network Work with the customer to define the feature access code parameters for use with

analog phones. Identify the transfer and call forwarding router patterns that suit the customers

requirements Identify the loopback interface and IP address that will be associated with the

CME service

20


Enable the CME feature on the router as follows:

Task 1: Configure the loopback interface ip address that will be used to signal SIP to AT&Ts FlexReach service.

interface Loopback0 description “ATT Provided SIP Signaling+Media Address” ip address <x.x.x.x> 255.255.255.255

Task 2: Enable CME feature and configure specific parameters. telephony-service max-ephones 166 ! Specify the maximum number of IP phones to be supported by the CME. max-dn 498 ! Specify the maximum number of extensions to be supported by the CME. ip source-address x.x.x.x port 2000 ! AT&T Assigned SIP Signaling address NAS in this example Loopback0 timeouts interdigit 5 ! Set the inter digit timeout to 5 seconds, default is 15 seconds. system message !! AT&T ! Configure the text message that will displayed on all phones time-format 24 ! Set the time to 24 hrs or 12 hrs am/pm format create cnf-files version-stamp 7960 Mar 05 2010 22:40:10 ! Create XML cnf files that are required for IP phones. The date and 6time stanp will be updated every time the phone is reset. keepalive 10 ! Specify the time interval in seconds (10 def=30), between keepalive messages sent by ip phones to the router. The router will un-register the phone after the keapalive timeout period. moh music-on-hold.au

! Define music-on-hold filename web admin system name cisco password cisco ! Specify the system administrator credentials, name and password. This allows web access to the CME router.

Task 3: Configure the firmware load files for the IP phones and SCCP gateways in the customer’s IPT network.

Important Information about Configuring Cisco Unified IP Phone Support

When configuring the load command:

- In Cisco Unified CME 7.0(1) and later versions, use the complete filename, including the file suffix, when you configure the load command for phone firmware versions later than version 8-2-2 for all phone types.

When configuring the load command for IP phones such as the Cisco Unified IP Phone 7906G, 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971G, configure only the filenames that are marked with an asterisk (*) in the table below.

21


Only SCCP phones can be configured as agent phones for Unified CCX 5.0 in Cisco Unified CME 4.2 and later versions. The Cisco VG224 Analog Phone Gateway and analog and SIP phones are supported as usual in Cisco Unified CME, however, not as Unified CCX agent phones.

Cisco Unified CME 7.0(1) Supported Firmware, Platforms and Voice Products can be found at the following url:http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme701spc.htm

telephony-service load 7905 P7905080001SCCP051117A!Select the firmware load file for the 7905 phoneload 7906 SCCP11.8-3-3S.loads! Select the firmware load file for the 7906 phoneload 7911 SCCP11.8-3-3S.loads! Select the firmware load file for the 7911 phoneload 7912 CP7912080003SCCP070409A.sbin! Select the firmware load file for the 7912 phoneload 7960-7940 P00308000500! Select the firmware load file for the 7960/7940 phoneload 7961 SCCP41.8-3-3S.loads! Select the firmware load file for the 7961 phoneload 7965 SCCP45.8-3-3S.loads! Select the firmware load file for the 7965 phoneload 7975 SCCP75.8-3-3S.loads! Select the firmware load file for the 7975 phone

Task 4: Configure the parameters that support supplementary services such as conference, transfer, and call forwarding.

telephony-service

max-conferences 8 gain 0 !define max number of 3 party G.711 conference and conference mixer gain as 6dB, 3db, 0db or -6dB (default) call-forward pattern .T ! Specify the route pattern for call forwarding transfer-system full-consult ! Set the call transfer system option to full-consult transfer-pattern .T ! Specify the route pattern for call transfer transfer-pattern 1.......... transfer-pattern 011T transfer-pattern 1.. transfer-pattern 2.. transfer-pattern 3.. transfer-pattern 4.. transfer-pattern 5.. transfer-pattern 6.. transfer-pattern 7.. transfer-pattern 8.. transfer-pattern 9..


telephony-service sdspfarm units 1 sdspfarm transcode sessions 16 sdspfarm tag 1 MTP0026cb5c16e1

22


no auto-reg-ephone max-ephones 20 max-dn 40 ip source-address 135.16.206.12 port 2000 calling-number initiator service phone rtcp timeouts interdigit 5 system message AT&T load 7905 CP7905080001SCCP051117A load 7906 SCCP11.8-3-3S.loads load 7911 SCCP11.8-3-3S.loads load 7912 CP7912080003SCCP070409A.sbin load 7960-7940 P00308000500 load 7961 SCCP41.8-3-3S.loads load 7965 SCCP45.8-3-3S.loads voicemail 7322162767 max-conferences 8 gain 0 moh flash:/CME/music-on-hold.au web admin system name cmeadmin password jets dn-webedit time-webedit transfer-system full-consult dss secondary-dialtone 9 directory last-name-first create cnf-files version-stamp 7960 Mar 05 2010 22:40:10

3.6 Dialing Restrictions

Class of Restriction (COR) is used to specify which incoming dial-peer can use an outgoing dial-peer to make a call. The configuration of COR involves the following four steps:

1. Define CoR custom name2. Associating each COR custom name with CoR list name that will be applied

to dial-peer associated with specific destination patterns3. Associating one or more CoR custom name with COR list name that will be

applied to ephone dn.4. Applying the CoR List to the dial-peer and ephone-dn

Configure COR as follows:

Task 1: Configure the names for the custom Class of Restriction that meet he customer’s requirements. For example, define CoR group name for the local, long-distance and N11 call flows.dial-peer cor custom !Define custom COR namename Localname Longdistancename N11

Task 2: Associate the COR names with the Class of Restriction list dial-peer that will be applied with dial-peers. Therefore, for each COR name define a corresponding COR list dial-peer.dial-peer cor list call-LongDistance ! Define List of Class of Restriction and name of Class of Restriction list (call-Longdistance)

23


member Longdistance !Specify that COR name Longdistance is a member of COR list call-Longditance.!dial-peer cor list call-Local member Local!dial-peer cor list call-N11 member N11

Task 3: Define a class of service Class of Restriction list that will be applied to ephone-dn users.dial-peer cor list COS3 member Local member N11!dial-peer cor list COS1 member Local member Longdistance member N11!dial-peer cor list COS4 member N11

Task 4: Apply the Class of Restriction list to the dial-peers

dial-peer voice 100 voip corlist outgoing call-LongDistance ! Set the outgoing Class of Restriction list translation-profile outgoing XlateCallingNumber destination-pattern 91.......... rtp payload-type nse 99 rtp payload-type nte 100 voice-class codec 1 voice-class sip asserted-id pai voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#1> dtmf-relay rtp-nte

Task 5: Apply the class of service Class of Restriction list to the ephone-dn. For example, phone uses associated with ephone-dn 9 will have the privilege to place local, long-distance, and N11 calls (i.e the dial-peer associated with those patterns). However, phone users associated with ephone-dn 12 will have privilege to place N11 and local calls only.

ephone-dn 1 dual-line number 2764 label 2764-1 description Sonus name Store #70 call-forward busy 7322162767 call-forward noan 7322162767 timeout 5

24


corlist incoming COS1! Apply the COR CoS COS1 list to the incoming calls from this phone

3.7 Phone Configuration

The CME router needs to be configured such that it allows phones to down load the firmware, place and receives calls. The process for defining and configuring phones in the CME router entails the following steps:

Define an ephone-dn Defining an ephone.

Load the appropriate phone software release that is compatible with the CME release refer to section 2.8 for configuration details.

3.7.1Ephone-dn

An ephone-dn represents a virtual line that connects a voice channel to a phone device on which a user can receive and make calls. Table 1 represents Cisco’s recommendation of the maximum ephone-dn that a platform can support. This number represents the maximum number of simultaneous call connections that can occur. The CCME creates an FXO virtual port and one or more dial-peer for each phone device so that it can process calls from or to the ephone-dn.

Before you configure an ephone-dn gather the following information:

Define the number of phone channels or lines per each phone (i.e. ephone-dn). The IPT offer recommends configuring all ephone-dns with two channels (i.e. dual-channel).Identify the class of service that the phone device /ephone-dn belong to. Example of features and services that a given phone device is going to support or is going to be participating include:

Pickup group Support call waiting Hunt group Call Park

Configure an ephone-dn as follows:

Task 1: Configure ephone-dn based on the customer’s phone profile requirements. The following is an example of two ephone-dn used in this project. The phone associated with ephne-dn 1 will have the following characteristics:

Will be part of the pickup group 20

25


Will have a dual line with a primary extension 2764 and secondary extension 2012 (i.e. primary+2 suffix)

Will use a different ring tone when receiving call on the secondary number. Will not be allowed to forward local calls A call to the primary number when it is busy will not hunt to the secondary line

ephone-dn 1 dual-line number 2764 secondary 1212 pickup-group 20 label 2764-1 name Store #70 call-forward busy 7322162767 call-forward noan 7322162767 timeout 5

3.7.2Ephone

An ephone is a logical representation of a telephone device on the CME rotuer. Thus, all phones (Wired, wireless, or analog) within the customer’s IPT network should have to be pre-configure on the CME router. An ephone also represents a port connected to a voice-mail system. Table 2 provides a list of maximum number of telephones (i.e. ephones) a given platform can support. Do the following before you configure an ephone.

Make an ordered list of the MC-Address of all IP phones within the customers IP network

Make an ordered list of the MAC-Address of all analog phones connected to a VG248, VG224, or ATA.

Identify the CME router providing telephony service to the ephone Identify the phone extension or ephone-dn that the ephone is associated to Identify the extension or ephone-dn for the numerous buttons on the phone (wired

IP phone only) Identify the phone buttons that need to support multiple extensions Determine the option for associating multiple ephone/extension to the button (e.g.

call waiting or overlay) Identify the ephone that use special ephone template

Configure an ephone as follows:

Task 1: Configure ephone based on the number of phones that a given CME is going to support. The following is an example of two ephone-dn used in this project. The phone associated with ephne-dn 1 will have the following characteristics:

ephone-dn 1 dual-line number 7322162764 label Business Sales description Business Sales name Business Sales call-forward busy 7322162767 call-forward noan 7322162767 timeout 5

26


corlist incoming COS1!!ephone-dn 2 dual-line number 7322162765 label Customer Service description Customer Service name Customer Service call-forward busy 7322162767 call-forward noan 7322162767 timeout 5ephone 1 device-security-mode none mac-address 000D.BCD8.0C50 type 7960 button 1:1!!!ephone 2 device-security-mode none mac-address 001A.2FF5.8E69 type 7940 button 1:2

3.8 Conference

Phone users should use the conference soft key or flash-hook to invoke the call conference features. Digital phone users can conference an active call by pressing the conference soft-key and dialing the new location and then commit the conference by pressing the conference soft-key again when appropriate. Analog phone users should also follow the same procedure but use the flash-hook instead of the conference soft-key.Before you configure a call conference feature on the CME gather the following information:

Define the call conference policy based on customer’s requirements. An example of a customer’s requirements can be some phones will only be allowed to conference local and/or long-distance calls.

Decide whether you want to disable call conference features on per phone or per phone template basis. The IP development team recommends to use the per phone template based feature control method.

Determine the maximum number of conference session required to support the customer’s requirement. Also, determine the DSP resource required to support them. Note that the NM-HDV network modules have five sockets or PVDM slots that each holds a 12-Channel PVDM (PVDM-12). Each PVDM-12 holds three T1 549 DSPs. Each DSP supports four channels. Use the Cisco DSP calculator at :

Define the conference initiator drop-off control policy. With CME 3.2 and later version introduced a feature that allows disconnecting or maintaining the conference call should the person who initiates a conference call hangs up.

Configure call conference feature involver the following two steps:

27


1. DSP Farm configuration 2. CME Configuration

Follow the following steps to configure a DSP farm for use with CME or other H323 GW.

Enable dsp farm on the voice card Define the Skinny Client Control protocol (SCCP)

Configure the CCM group Assign priority to the group Associate the group with a DSP farm profile Set keepalive, switchback, and switchover parameters. Use defaults.

Configure the DSP farm profile Specify the maximum f Transco ding sessions per dsp farm Select the local sscp interface for use with the dsp farm Specify the codec types that will be supported by the profile Associate sccp with the dsp farm profile

Configure the CCME to act as the DSP farm host Set the ccm router to receive ip phone messages over the CCME router’s ip

address Set sccp server to use the max number of dsp farms Set the cme router to allow for a maximum number of g711 and g729 transcoded

sessions Tag and define dsp farm units for cme router registry

Task1: Enable dsp services on the slot housing the NM-HDV or PVDM farm

voice-card 0 dspfarm dsp services dspfarm ! enables DSP-farm services on the NM-HDV or PVDM

Task2: Select the local interface that the SCCP applications (transcoding or conferencing) should use to register with CCME or CCM. Use the network management or the voice LAN virtual-interface.

sccp local GigabitEthernet0/0.1sccp ccm 10.30.10.1 identifier 2851 version 7.0 ! Specify the CCME or CCM address. This is used to associate the SCCP CCME ip address with the cmm groupsccp ! Enables SCCP and its associated transcoding and conferencing application

Task3: Define the transcoding profile for the DSP farm. There is no need to define the conference profile.

28


dspfarm profile 1 transcode ! Enabled profile for transcoding and enters the sp farm profile configuration mod. Select a unique number to identify the profile, range 1-65535 for IPT use 2 codec g711ulaw ! specify the code types supported by the dsp farm profile codec g711alaw codec g729ar8 codec g729abr8 codec g729br8 codec g729r8 maximum sessions 16 ! Specify the max number of sessions that are supported by the profile. Range 0-X. Def=0. The value for X is determined at run time depending on the number of resources available with the resource provider. associate application SCCP ! associates sccp with the profile

Task4: Create CCME group. Only one group is required.

sccp ccm group 1 ! create ccme group number range [1-65535]. For IPT use 1 associate ccm 2851 priority 1 ! Specify the priority of the CCME router within the ccme group. Default is 1 because only one ccme group is possible. associate profile 1 register MTP0026cb5c16e1 ! Associate DSP farm profile with the ccme group. Use the MAC address of the sccp client interface with the “MTP” prefix added. keepalive retries 5 ! adjust the keepalive to 5 range is 1-32 default is 3

Task5: Configure the CCME router to act as the DSP farm host.

Telephoney-service ! enter the telephony service configuration modesdspfarm units 1 ! specifies the max number of dsp farm that are allowed to be registered to the sccp server. For IPT set it to 1. Range 0-5 def =0 sdspfarm transcode sessions 16 ! Specify the max number of transcode sessions for g729 allowed by the ccme router. Range 0-128 def=0 sdspfarm tag 1 MTP0026cb5c16e1! Permit a dspfarm unit to be registered to ccme and associate it with a sccp client interface mac address. Tag number range 1-5 and device name is the mac address of the sccp client interface, with the mtp prefix addedmax-conferences 8 gain 0 !define max number of 3 party G.711 conference and conference mixer gain as 6dB, 3db, 0db or -6dB (default)

Task6: Activate the trans-coder profile

dspfarm profile 1 transcode no shut

Task 7: Configure the conference imitator drop-off control parameters per ephones. Without this command the conference will be disconnected if the conference initiator hangs up.

ephone 1 device-security-mode none mac-address 000D.BCD8.0C50 speed-dial 1 103 label "Speed-dial" type 7960 keep-conference! Do not disconnect conference when conference initiator hangs-up. This will connect the remaining parties using call transfer. button 1:1 2:7 3:8

29


!ephone 2 device-security-mode none mac-address 001A.2FF5.8E69 speed-dial 1 103 label "Speed-dial" type 7940 keep-conference endcall! Do not disconnect conference when conference initiator hangs-up. This will connect remaining parties using call transfer. If the endcall option is selected, pressing the EndCall soft-key on the IP phone terminates the conference but keeps the remaining parties connected (via transfer).If the endcall option is not selected, pressing EndCall terminates the conference and disconnects all parties. button 1:2

Transcoding Resource

CME uses a software-based audio mixing of G711 audio streams to support three-party conferencing. CME supports conferencing between G711 and G729 calls by transcoding the G729 call to G711 through the use of DSP resource. The DSP resources are therefore used by the CME to indirectly support conferencing functionality. Transcoding DSP resources are only required to support conferencing in a mixed mode environment which is multi-site with distributed CME routers topology. Transcoding DSP resources are not required for a single-site with local CME router topology since all calls are G711.

Order the appropriate Voice Over service feature package with additional DSPs to accommodate mixed mode conferencing requirements for each CME site. The DSP resource must reside in the local CME router.


sccp local GigabitEthernet0/1sccp ccm 10.30.10.1 identifier 2851 version 7.0 Customer LAN Addresssccp!sccp ccm group 1 associate ccm 2851 priority 1 associate profile 1 register MTP0026cb5c16e1 keepalive retries 5!dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729br8 codec g729r8 maximum sessions 16 associate application SCCP!

Call conference feature design considerations.

Define the call conference policy based on some phones will only be allowed to conference local and/or long-distance calls.

30


Decide whether you want to disable call conference features on per phone or per phone template basis. The recommendation is to use the per phone template based control method.

Determine the maximum number of conference session required to support the site requirements. Determine the percentage of conference participant that require transcoding to join the conference. Based on this percentage determine the DSP resource required to support them. Use the Cisco DSP calculator at : http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl

Define the conference initiator drop-off control policy. CME has a feature that allows disconnecting or maintaining the conference call should the person who initiates a conference call hangs up.

3.9 VoIP Connectivty to AT&T IP Flexible Reach

The following configuration is required to configure the router for SIP signaling with the AT&T Border Elements. In this configuration, the loopback 0 interface is used for all SIP signaling traffic. Loopback 0 must be a registered IP address.

voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 3600 min 3600 asserted-id pai

sip-ua retry invite 2

3.10 Codec configuration

A voice class codec can be used to provide a list of codecs with preferences which the dial peers will refer to. The codec with the lower preference number has the highest preference. For example, preference 1 has a higher preference than preference 2.

voice class codec 1 codec preference 1 g729br8 bytes 30 codec preference 2 g729r8 bytes 30 codec preference 3 g711ulaw

31

http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl


3.11 SIP Profile

voice class sip-profiles 1 request INVITE sdp-header Audio-Attribute add "a=ptime:30" response 200 sdp-header Audio-Attribute add "a=ptime:30" request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:732216\1@\2>"

** For forwarded calls from CISCO CME user to PSTN (out to AT&T’s IP Flex-reach service) some AT&T serviced areas require that the SIP Diversion header contain the full 10-digit DID number of the forwarding party. In this configuration guide the assumption was made that a typical customer will utilize extension numbers (4-digit assignments in this example) and map 10-digit DID number using CISCO UCME translation pattern. Because we use 4-digit extensions on our CISCO UCME IP phones it is necessary to expand the 4-digit extension, included in the Diversion: header of a forwarding INVITE message, to its full 10-digit DID number when the IP phone is set to call-forward. The requirement to expand the Diversion-Header has been achieved by the use of a SIP profile in CISCO CME.

**This Diversion header example assumes all TNs at the site begin with 732216..., if this is not the case and some phones have different NPA-NXX TNs then a full static 10 digit TN could be used instead of 732216...

3.12 Privacy with SIP (Optional)

The CME router with the SIP trunk connectivity to AT&T’s Flexreach service can be provided with the option of sending calls with privacy enabled. The called party will see these calls with caller id indicating Anonymous, Unknown Number or blocked (depending on the PBX/PSTN network at the destination).

This feature is invoked with the use of the following commands –

asserted-id pai – This is explained above. We will make it the default configuration starting with the introduction of 12.4.15T6a IOS.

privacy id – This is used under the voice service voip, sip section of the configuration and applies to all dial-peers.

Example –

voice service voip sip privacy id

In the invite sent from the CME router to the IP BE –

1. The FROM header is changed to From: <sip:[email protected]> 2. P-Asserted-Identity message header is added with privacy id field

P-Asserted-Identity: sip:[email protected]

32


Privacy: id

Per RFC 3325, The AT&T IP BE removes the P-Asserted-identity header and sends the invite to the destination with the following -

From: "Anonymous"<sip:[email protected]:5060>Contact: <sip:[email protected]:5060;transport=udp>

** Note – If this option is used all registered ephones will be set to privacy.

3.13 Calling/Called Party Number Translation Configuration

This section defines the voice translation configuration to support 4-digit DNs on the IP Phones. In this example the IP Phones will use 3143323765 as the calling party number for all outbound calls, optionally voice translation rule 1 can be used to match on a per DN basis instead.

Defining a voice translation rule that translates an incoming or outgoing call number based on a match rule criteria. This process involves three steps:

1. Configure the voice translation rule with the desired match and replace criteria2. Configure a translation profile based for each translation rule on whether the

ruleswill be applied to incoming(calling) or outgoing(called).3. Apply the translation profile to the desired voip dial-peer.

voice translation-rule 1 (rule per DN) rule 1 /2764/ /7322162764/ rule 2 /2765/ /7322162765/!voice translation-rule 3 (global rule for all outbound calls) rule 1 /$.*$/ /3143323765/

voice translation-rule 4 (strips 9 off dialed number and sends 11 called digits to Flex IPBE) rule 1 /^9$9..$/ /\1/ rule 2 /^9$4..$/ /\1/ rule 3 /^9$[2-9].........$/ /\1/ rule 4 /^9$1[2-9].........$/ /\1/ rule 5 /^9$011............$/ /\1/

voice translation-profile XlateCallingNumber translate calling 3 translate called 4

voice translation-rule 2 rule 1 /7322162764/ /2764/ rule 2 /7322162765/ /2765/!

33


voice translation-profile HopOnXlations translate called 2

Note on Incoming Call Routing

Note that for virtual telephone numbers, AT&T will send 10 digits. For non virtual DIDs, AT&T will send 7 digits or less. The customer may request the exact number of digits that they want to receive. In this example to 10-digit called number is translated to a 4-digit DN.

3.14 Outbound Calls to AT&T

dial-peer voice 100 voip description Outgoing Dial Peer to Border Element #1 for US Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber destination-pattern 91.......... voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#1> dtmf-relay rtp-nte

dial-peer voice 101 voip description Outgoing Dial Peer to Border Element #2 for US Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber preference 1 destination-pattern 91.......... voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#2> dtmf-relay rtp-nte

dial-peer voice 102 voip description Outgoing Dial Peer to Border Element #1 for International Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber preference 1 destination-pattern 9011T voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1

34


session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#1> dtmf-relay rtp-nte

dial-peer voice 103 voip description Outgoing Dial Peer to Border Element #2 for International Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber preference 1 destination-pattern 9011T voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#2> dtmf-relay rtp-nte

dial-peer voice 104 voip description Outgoing Dial Peer to Border Element #1 for Local Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber preference 1 destination-pattern 9[2-9]11 voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#1> dtmf-relay rtp-nte

dial-peer voice 105 voip description Outgoing Dial Peer to Border Element #2 for Local Calls corlist outgoing COS1 translation-profile outgoing XlateCallingNumber preference 1 destination-pattern 9[2-9]11 voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class sip profiles 1 session protocol sipv2 session target ipv4: <AT&T IP Border Element IP Addr#2> dtmf-relay rtp-nte

dial-peer hunt 1 <required to round robin between Border Elements #1 and #2>

35


dial-peer hunt 1 indicates that dial peers will be selected by the following criteria 1) longest match in phone number,2) explicit preference and 3) random selection. This creates a load balancing across multiple dial-peers with the same destination-pattern and the same preference.

3.15 Inbound Calls from AT&T

dial-peer voice 400 voip translation-profile incoming HopOnXlations rtp payload-type nse 99 rtp payload-type nte 100 voice-class codec 1 session protocol sipv2 voice-class sip profiles 1 incoming called-number 732216.... <EnterpriseDNs> (assumes 10-digits being sent by Flex IPBE) dtmf-relay rtp-nte

4 Unity Express Configuration

This section provides the configuration guidelines for the steps needed to configure Unity Express (CUE) voicemail on the CME router.

Configure the router housing the CUE as follows:

Task 1: Configure an ip address on service engine interface. Use a 30 bit mask address.

interface Integrated-Service-Engine1/0 0 ! This is the router slot hosting the Cisco unity express network module

ip address 10.10.33.1 255.255.255.252 ! Assign the odd part of the host ip address to interface. The router uses this interface to communicate with the CUE service module. service-module ip address 10.10.33.2 255.255.255.252 ! Assign the service module the even part of the host address IP address service-module ip default-gateway 10.10.33.1 !set the default gateway to the address of service-engine

no keepalive

Task 2: Configure a SIP dial peer that points to the CUE network module for Voicemail and/or Auto Attendandt feature. This is required because the CUE can only communicate with the CME using SIP protocol.

dial-peer voice 9999 voip ! SIP dial-peer pointing to CUE. CUE supports only SIP calls description NM-CUE Unity Express VM destination-pattern 7322162767 <CUE Pilot TN>! session protocol sipv2 ! Enable SIP protocol session target ipv4:10.10.33.2 ! CUE ip address dtmf-relay sip-notify codec g711ulaw ! CUE supports only G711 calls

36


no vad ! Disable VAD

CUE AA dial-peer configuration

dial-peer voice 500 voip ! SIP dial-peer pointing to CUE AA. CUE supports only SIP calls description NM-CUE Unity Express destination-pattern 7322162766 !<CUE AA TN>! session protocol sipv2 session target ipv4:10.10.33.2! CUE ip address dtmf-relay sip-notify codec g711ulaw ! CUE supports only G711 calls no vad!dial-peer voice 501 voip ¡ Inbound SIP dial-peer for CUE AA calls received from FlexReach voice-class codec 1 rtp payload-type nse 99 rtp payload-type nte 100 voice-class codec 1 voice-class sip profiles 1 session protocol sipv2 incoming called-number 7322162766!

Task 3: Make a list of individual and group phone user’s extensions that the customer wants the calls to be forwarded to the standard mail box when the line is busy or not answering. Configure the ephone-dn to forward the calls to the voice mailbox when the line is busy or not answering.

ephone-dn 1 dual-line number 2764 secondary 1212 pickup-group 20 label 2764-1 name Store #70 call-forward busy 7322162767 call-forward noan 7322162767 timeout 5 ! Forwards the call to voice maibox after 5 seconds

Task 4: Configure MWI dn. This will allow the CME to relay SIP calls from the CUE to the telephone affected to change the status of the telephone’s message waiting indication light. The CUE places a MWI notification call by appending the phones extension to the MWI extension (e.g. 80018026). The CME will route the call to ephone-dn 80 in this example. EPhone-dn 80 will accept the call and switches on the MWI light for the phone extension (e.g. 8026).

ephone-dn 3 number 8000.... ! 8000 is the MWI on extension configured on the CUE followed by 4 dots representing the length of the local phone extension. The number of dots must be equivalent to the extension length of the customer’s private dial-plan. Also make sure the MWI number matches the extension configured on the CUE. mwi on!

37


ephone-dn 4 number 8001.... mwi off

Task 5: Add the voice mail box extension to the Telephony service CCME application

Telephony-service voicemail 7322162767

38


5 Troubleshooting

CME1#show inventory

NAME: "2851 chassis", DESCR: "2851 chassis"PID: CISCO2851 , VID: V05 , SN: FTX1343AJ6T

NAME: "WAN Interface Card - Serial 2T on Slot 0 SubSlot 0", DESCR: "WAN Interface Card - Serial 2T"PID: WIC-2T= , VID: 1.0, SN: 23739928

NAME: "PVDMII DSP SIMM with three DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSPSIMM with three DSPs"PID: PVDM2-48 , VID: V01 , SN: FOC13380KMS

NAME: "Integrated Service Engine for Modular and Integrated Services Routers on Slot 1", DESCR: "Integrated Service Engine for Modular and Integrated Services Routers"PID: NME-CUE , VID: V01 , SN: FOC121434Q4

CME1#

alias exec call show call active voice | in PeerIdalias exec rtp show voip rtp conalias exec codec show call active voice | in Codalias exec sccp show sccp connalias exec connect show voice call summ | in CON

sh versh runsh flash:sh call app voice summ

If the calls still fail - would need the below debugs:

deb voip ivr scriptdeb vpm sigdeb voip dialpeer inoutdeb voip ccapi inout

39


Follow the following steps when verifying the configuration of the CME in the customer’s IPT network.Use the procedures outlined in the troubleshooting guidelines Section to resolve any problem encountered during the site acceptance test.

1) Collect the following information before accessing the CME: List of ephone and ephone-dn that will be used to test the CME feature (at least one

analog phone, IP phone and wireless phone if applicable). The total number of configured ephones Telephone extension and/or range, dhcp ip address pool range, voice vlan, access

switch ip address, and access switch ports associated with the phones that will be used for testing the CME feature.

List of CME features such as conferencing, huntgroup .. List of ip address of all other IPT servers that interwork with the CME. Examples

include: voice mail system, dhcp server, network gatekeeper, and IP PBX at remote locations (CME, Call manager or IOS gateways).

Dial plan for handling outgoing and incoming calls. Telephone number associated with circuits that connect the CME to the PSTN (e.g.

FXO, PRI or T1). Onsite personnel that have access to the phones that will be used for testing the CME

features.

Access the CME and verify that the CME is running the IPT supported IOS release and CME version Use the sh telephony-service all command to verify the CME feature is enabled on the router. Note this command also displays the whole CME configurationCME1#sh telephony-service allCONFIG (Version=7.0(1))=====================Version 7.0(1)Cisco Unified Communications Manager ExpressFor on-line documentation please see:http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html

ip source-address 135.16.206.13 port 2000no auto-reg-ephoneload 7905 CP7905080001SCCP051117Aload 7906 SCCP11.8-3-3Sload 7911 SCCP11.8-3-3Sload 7912 CP7912080003SCCP070409A.sbinload 7960-7940 P00308000500load 7961 SCCP41.8-3-3Sload 7965 SCCP45.8-3-3Sload 7975 SCCP75.8-3-3Smax-ephones 20max-dn 40max-conferences 8 gain 0dspfarm units 1dspfarm transcode sessions 16dspfarm 1 MTP0026cb5c16e1

40


conference softwareprivacyno privacy-on-holdhunt-group report delay 1 hourshunt-group logout DNDmax-redirect 5voicemail 7322162767cnf-file location: system:cnf-file option: PER-PHONE-TYPEnetwork-locale[0] US (This is the default network locale for this box)network-locale[1] USnetwork-locale[2] USnetwork-locale[3] USnetwork-locale[4] USuser-locale[0] US (This is the default user locale for this box)user-locale[1] USuser-locale[2] USuser-locale[3] USuser-locale[4] USsrst mode auto-provision is OFFsrst ephone template is 0srst dn template is 0srst dn line-mode singlephone service rtcp 1moh flash:/CME/music-on-hold.autime-format 12date-format mm-dd-yytimezone 0 Greenwich Standard Timesecondary-dialtone 9transfer-pattern 1..........transfer-pattern .Tkeepalive 30 auxiliary 30timeout interdigit 5timeout busy 10timeout ringing 180timeout transfer-recall 0timeout ringin-callerid 8timeout night-service-bell 12caller-id name-only: enablesystem message AT&T Store #70web admin system name cmeadmin password jetsweb admin customer name Customeredit DN through Web: enabled.edit TIME through web: enabled.Log (table parameters): max-size: 150 retain-timer: 15create cnf-files version-stamp 7960 Mar 08 2010 15:36:00transfer-system full-consult dsstransfer-digit-collect new-calllocal directory service: enabled.Extension-assigner tag-type ephone-tag.

41


ephone-dn 1 dual-linenumber 2764name Store #70description Sonuslabel 2764-1preference 0 secondary 9huntstopno huntstop channelcall-forward busy 7322162767call-forward noan 7322162767 timeout 5call-waiting beepcor incoming COS1

ephone-dn 2 dual-linenumber 2765name Store #70description Sonuslabel 2765-1preference 0 secondary 9huntstopno huntstop channelcall-forward busy 7322162767call-forward noan 7322162767 timeout 5call-waiting beep

ephone-dn 3number 8000..........preference 0 secondary 9huntstopcall-waiting beepmwi on

ephone-dn 4number 8001..........preference 0 secondary 9huntstopcall-waiting beepmwi offephone-dn 5 dual-linenumber 7322162766name Store #70description Sonuslabel 2766-1preference 0 secondary 9huntstopno huntstop channelcall-forward busy 7322162767call-forward noan 7322162767 timeout 5call-waiting beep

42






ephone-dn 10 dual-linenumber 3764name Store #70description HIPCS/NSNlabel 3764-1

43


preference 0 secondary 9huntstopno huntstop channelcall-waiting beep

ephone-dn 11 dual-linenumber 3765name Store #70description HIPCS/NSNlabel 3765-1preference 0 secondary 9huntstopno huntstop channelcall-waiting beep

Number of Configured ephones 7 (Registered 6)ephone 1Device Security Mode: Non-Securemac-address 000D.BCD8.0C50type 7960button 1:1 2:7 3:8speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US

ephone 2Device Security Mode: Non-Securemac-address 001A.2FF5.8E69type 7940button 1:2speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conference endcallconference drop-mode neverconference add-mode allconference admin: No

44


privacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 3Device Security Mode: Non-Securemac-address 0019.567E.8C89type 7961button 1:6speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 4Device Security Mode: Non-Securekeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 5Device Security Mode: Non-Securemac-address 0013.C326.841Ftype 7905button 1:9keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: No

45


Preferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 6Device Security Mode: Non-Securemac-address 001F.CA34.B2BBtype 7965button 1:10keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 7Device Security Mode: Non-Securemac-address 0021.5553.8707type 7975button 1:11keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!voice-port 50/0/1 station-id number 2764 station-id name Store #70 timeout interdigit 5 timeout ringing 5!

46


voice-port 50/0/2 station-id number 2765 station-id name Store #70 timeout interdigit 5 timeout ringing 5!voice-port 50/0/3 station-id number 80000000000000 timeout interdigit 5!voice-port 50/0/4 station-id number 80010000000000 timeout interdigit 5!voice-port 50/0/5 station-id number 7322162766 station-id name Store #70 timeout interdigit 5 timeout ringing 5!voice-port 50/0/6 station-id number 2162768 station-id name Store #70 timeout interdigit 5 timeout ringing 5!voice-port 50/0/7 station-id number 2764 station-id name Store #76 timeout interdigit 5 timeout ringing 5voice-port 50/0/8 station-id number 2764 station-id name Store #70 timeout interdigit 5 timeout ringing 5!voice-port 50/0/9 station-id number 2766 station-id name Store #70 timeout interdigit 5 timeout ringing 5!voice-port 50/0/10 station-id number 3764 station-id name Store #70 timeout interdigit 5!voice-port 50/0/11 station-id number 3765 station-id name Store #70 timeout interdigit 5!

47


dial-peer voice 20001 pots destination-pattern 2764$ huntstop call-forward busy 7322162767 call-forward noan 7322162767 corlist incoming COS1 progress_ind setup enable 3 port 50/0/1dial-peer voice 20002 pots destination-pattern 2765$ huntstop call-forward busy 7322162767 call-forward noan 7322162767 progress_ind setup enable 3 port 50/0/2

dial-peer voice 20003 pots destination-pattern 8000.......... huntstop progress_ind setup enable 3 port 50/0/3


dial-peer voice 20005 pots destination-pattern 7322162766$ huntstop call-forward busy 7322162767 call-forward noan 7322162767 progress_ind setup enable 3 port 50/0/5



48




dial-peer voice 20010 pots destination-pattern 3764$ huntstop progress_ind setup enable 3 port 50/0/10

dial-peer voice 20011 pots destination-pattern 3765$ huntstop progress_ind setup enable 3 port 50/0/11tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xmltftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xmltftp-server system:/its/united_states/7960-font.xml alias English_United_States/7920-font.xmltftp-server system:/its/united_states/7960-dictionary.xml alias English_United_States/7960-dictionary.xmltftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7960-kate.xmltftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7920-kate.xmltftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_States/SCCP-dictionary.xmltftp-server system:/its/vrf1/XMLDefault7960.cnf.xml alias SEP000DBCD80C50.cnf.xmltftp-server system:/its/vrf1/XMLDefault7940.cnf.xml alias SEP001A2FF58E69.cnf.xmltftp-server system:/its/vrf1/XMLDefault7961.cnf.xml alias SEP0019567E8C89.cnf.xmltftp-server system:/its/vrf1/XMLDefault7905.cnf.xml alias SEP0013C326841F.cnf.xmltftp-server system:/its/vrf1/XMLDefault7965.cnf.xml alias SEP001FCA34B2BB.cnf.xmltftp-server system:/its/vrf1/XMLDefault7975.cnf.xml alias SEP002155538707.cnf.xmltftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnftftp-server system:/its/vrf1/XMLDefault.cnf.xml alias XMLDefault.cnf.xmltftp-server system:/its/vrf1/ATADefault.cnf.xml alias ATADefault.cnf.xml

Gather the customer’s dial plan. Make sure that the customer’s private dial plan is properly provisioned on the gatekeeper.

49


Use the show telephony-service ephone command to verify the number of ephones configured and how many are registered with the CME.

CME1#sh telephony-service ephoneNumber of Configured ephones 7 (Registered 6)ephone 1Device Security Mode: Non-Securemac-address 000D.BCD8.0C50type 7960button 1:1 2:7 3:8speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 2Device Security Mode: Non-Securemac-address 001A.2FF5.8E69type 7940button 1:2speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conference endcallconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 3Device Security Mode: Non-Securemac-address 0019.567E.8C89type 7961button 1:6

50


speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 4Device Security Mode: Non-Securekeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 5Device Security Mode: Non-Securemac-address 0013.C326.841Ftype 7905button 1:9keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 6

51


Device Security Mode: Non-Securemac-address 001F.CA34.B2BBtype 7965button 1:10keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!ephone 7Device Security Mode: Non-Securemac-address 0021.5553.8707type 7975button 1:11keepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US

Use the show telephony-ser ephone and ephone-dn command to verify the configuration of a subset of ephone-dn and the respective ephones.

CME1#sh telephony-ser ephone | beg ephone 1ephone 1Device Security Mode: Non-Securemac-address 000D.BCD8.0C50type 7960button 1:1 2:7 3:8speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: No

52


Preferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: Nouser-locale USnetwork-locale US!CME1#sh telephony-ser ephone-dn | beg ephone-dn 1ephone-dn 1 dual-linenumber 2764name Store #70description Sonuslabel 2764-1preference 0 secondary 9huntstopno huntstop channelcall-forward busy 7322162767call-forward noan 7322162767 timeout 5call-waiting beepcor incoming COS1

Also use the show ephone-dn command to verify the operational state and setting (such inter-digit timeout Call Disconnect Time ect) related to an ephone-dn.

CME1#sh ephone-dn 150/0/1 CH1 IDLE CH2 IDLE

EFXS 50/0/1 Slot is 50, Sub-unit is 0, Port is 1 Type of VoicePort is EFXS 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 0 dB Echo Cancellation is enabled Echo Cancellation NLP mute is disabled Echo Cancellation NLP threshold is -21 dB Echo Cancel Coverage is set to 8 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

53


Playout-delay Fax is set to 300 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 15 s Interdigit Time Out is set to 5 s Call Disconnect Time Out is set to 60 s Ringing Time Out is set to 5 s Wait Release Time Out is set to 30 s Companding Type is u-law Region Tone is set for US Station name Store #70, Station number 2764

Caller ID Info Follows: Standard BELLCORE Translation profile (Incoming): Translation profile (Outgoing):

Digit Duration Timing is set to 100 ms

Use the show telephony-service dial-peer command to verify that the CME has created and associated a dial-peer with a virtual port. For example, the CMPE will create two pots dial-peers for ephone-dn 1.

CME1#sh telephony-service dial-peer

dial-peer voice 20001 pots destination-pattern 2764$ huntstop call-forward busy 7322162767 call-forward noan 7322162767 corlist incoming COS1 progress_ind setup enable 3 port 50/0/1



dial-peer voice 20004 pots destination-pattern 8001..........

54


huntstop progress_ind setup enable 3 port 50/0/4






dial-peer voice 20010 pots destination-pattern 3764$ huntstop progress_ind setup enable 3 port 50/0/10

dial-peer voice 20011 pots destination-pattern 3765$ huntstop

55


progress_ind setup enable 3 port 50/0/11

CME1#

Use the show ephone reg command to confirm the customer phones are properly configured and registered with the CME, and are assigned the desired extension.

CME1#sh ephone reg | beg ephone-1ephone-1[0] Mac:000D.BCD8.0C50 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 11/9mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8IP:10.30.10.82 50127 Telecaster 7960 keepalive 14454 max_line 6 available_line3button 1: dn 1 number 2764 CH1 IDLE CH2 IDLEbutton 2: dn 7 number 2764 CH1 IDLE CH2 IDLEbutton 3: dn 8 number 2764 CH1 IDLE CH2 IDLEspeed dial 1:103 Speed-dialPreferred Codec: g711ulaw

Identify the mac address of the phone using command:

CME1#sh telephony-service ephone | beg ephone 1ephone 1Device Security Mode: Non-Securemac-address 000D.BCD8.0C50type 7960button 1:1 2:7 3:8speed-dial 1 103 label Speed-dialkeepalive 30 auxiliary 30multicast-mohmax-calls-per-button 8busy-trigger-per-button 0Always send media packets to this router: NoPreferred codec: g711ulawkeep-conferenceconference drop-mode neverconference add-mode allconference admin: Noprivacy: Yesprivacy button: No56


user-locale USnetwork-locale US!Turn on debug ephone reg for the specific ephone – CME1#debug ephone register mac-address 000D.BCD8.0C50EPHONE registration debugging is enabled for phone 000D.BCD8.0C50

Clear the logCME1#clear loggClear logging buffer [confirm]yCME1#

Reset the phone –

CME1#config tEnter configuration commands, one per line. End with CNTL/Z.CME1(config)#ephone 1CME1(config-ephone)#resetreseting 000D.BCD8.0C50

Display the log message using the sho log command:

CME1(config-ephone)#000289: May 5 16:09:46.471: ephone-1[0/3]:UnregisterMessage after Reset/Restart sent000290: May 5 16:09:46.471: ephone-1[0/3][SEP000DBCD80C50]:Phone Unregistered on socket [3] SEP000DBCD80C50000291: May 5 16:09:46.475: ephone-1[0/3]:UnregisterAck sent on socket [3] (5/6/21)000292: May 5 16:09:46.475: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP000DBCD80C50 IP:10.30.10.82 Socket:3 DeviceType:Phone has unregistered normally.

CME1#show logg

Syslog logging: enabled (0 messages dropped, 1 messages rate-limited, 0 flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.

No Inactive Message Discriminator.

57


Console logging: level debugging, 384 messages logged, xml disabled, filtering disabled Monitor logging: level debugging, 0 messages logged, xml disabled, filtering disabled Buffer logging: level debugging, 384 messages logged, xml disabled, filtering disabled Logging Exception size (4096 bytes) Count and timestamp logging messages: disabled Persistent logging: disabled

No active filter modules.

ESM: 0 messages dropped

Trap logging: level informational, 295 message lines logged

Log Buffer (100000 bytes):

000289: May 5 16:09:46.471: ephone-1[0/3]:UnregisterMessage after Reset/Restart sent000290: May 5 16:09:46.471: ephone-1[0/3][SEP000DBCD80C50]:Phone Unregistered on socket [3] SEP000DBCD80C50000291: May 5 16:09:46.475: ephone-1[0/3]:UnregisterAck sent on socket [3] (5/6/21)000292: May 5 16:09:46.475: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP000DBCD80C50 IP:10.30.10.82 Socket:3 DeviceType:Phone has unregistered normally.000293: May 5 16:13:19.587: New Skinny socket accepted [2] from 0, sub 1 (9 active)000294: May 5 16:13:19.587: sin_family 2, sin_port 50095, in_addr 10.30.10.82000295: May 5 16:13:19.587: skinny_add_socket 2 10.30.10.82 50095000296: May 5 16:13:19.595: %IPPHONE-6-REG_ALARM: 25: Name=SEP000DBCD80C50 Load=8.1(1.0) Last=Initialized000297: May 5 16:13:19.595: ephone-(1)[10] StationRegisterMessage (5/6/21) from 10.30.10.82000298: May 5 16:13:19.595: ephone-(1)[10] Register StationIdentifier DeviceName SEP000DBCD80C50000299: May 5 16:13:19.595: ephone-(1)[10] StationIdentifier Instance 1 deviceType 758


000300: May 5 16:13:19.595: ephone-1[0/3]:stationIpAddr 10.30.10.82000301: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:maxStreams 0000302: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:From Phone raw protocol Ver 0x8560000B000303: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:protocol Ver 0x8560000B000304: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:phone-size 57240 dn-size 784000305: May 5 16:13:19.595: ephone-(1) Allow any Skinny Server IP address 135.16.206.13000306: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:Found entry 0 for 000DBCD80C50000307: May 5 16:13:19.599: ephone-1[0/3][SEP000DBCD80C50]:socket change 3 to 10000308: May 5 16:13:19.599: ephone-1[0/3][SEP000DBCD80C50]:DisAssociate: Closed socket 3 for unregistered phone000309: May 5 16:13:19.599: ephone-1[0/-1][SEP000DBCD80C50]:FAILED: CLOSED old socket -1000310: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:***Force device subtype to 0000311: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:phone SEP000DBCD80C50 re-associate OK on socket [10]000312: May 5 16:13:19.599: %IPPHONE-6-REGISTER: ephone-1:SEP000DBCD80C50 IP:10.30.10.82 Socket:10 DeviceType:Phone has registered.000313: May 5 16:13:19.599: Phone 0 socket 10000314: May 5 16:13:19.599: Skinny Local IP address = 135.16.206.13 on port 2000

000315: May 5 16:13:19.599: Skinny Phone IP address = 10.30.10.82 50095000316: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:Signal protocol ver 9 to phone with ver 11000317: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:Date Format M/D/Y000318: May 5 16:13:19.599: ephone-1[0/10]:RegisterAck sent to sockettype ephone socket 10: keepalive period 30 use sccp-version 9000319: May 5 16:13:19.599: ephone-1[0/10]:CapabilitiesReq sent000320: May 5 16:13:19.607: ephone-1[0/10]:HeadsetStatusMessage

59


000321: May 5 16:13:19.639: ephone-1[0/10]:MediaPathCapabilitiesMessage000322: May 5 16:13:19.643: ephone-1[0/10]:MediaPathEventMessage000323: May 5 16:13:19.643: ephone-1[0/10]:MediaPathCapabilitiesMessage000324: May 5 16:13:19.647: ephone-1[0/10]:MediaPathEventMessage000325: May 5 16:13:19.647: ephone-1[0/10]:CapabilitiesRes received000326: May 5 16:13:19.647: ephone-1[0/10][SEP000DBCD80C50]:Caps list 8WideBand_256K 120 ms, is_mtp 0G711Ulaw64k 40 ms, is_mtp 0G711Alaw64k 40 ms, is_mtp 0G729AnnexB 60 ms, is_mtp 0G729AnnexAwAnnexB 60 ms, is_mtp 0G729 60 ms, is_mtp 0G729AnnexA 60 ms, is_mtp 0Unrecognized Media Type 257 4 ms, is_mtp 0

000327: May 5 16:13:19.659: New Skinny socket accepted [2] from 0, sub 1 (9 active)000328: May 5 16:13:19.659: sin_family 2, sin_port 50096, in_addr 10.30.10.82000329: May 5 16:13:19.659: skinny_add_socket 2 10.30.10.82 50096000330: May 5 16:13:19.659: ephone-1[0/10]:MediaPathEventMessage000331: May 5 16:13:19.807: ephone-1[0/10]:MediaPathEventMessage000332: May 5 16:13:19.807: ephone-1[0/10]:ButtonTemplateReqMessage000333: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:StationButtonTemplateReqMessage set max presentation to 6000334: May 5 16:13:19.807: ephone-1[0/10]:CheckAutoReg000335: May 5 16:13:19.807: ephone-1[0/10]:AutoReg is disabled000336: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:Setting 3 lines 1 speed-dials on phone (max_line 6)000337: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:First Speed Dial Button location is 4 (0)000338: May 5 16:13:19.807: ephone-1[0/10]:ButtonTemplate lines=3 speed=1 buttons=6 offset=0

60


000339: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:ButtonTemplate buttonCount=6 totalButtonCount=6 buttonOffset=0000340: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:Configured 3 speed dial buttons000341: May 5 16:13:19.811: ephone-1[0/10]:StationSoftKeyTemplateReqMessage000342: May 5 16:13:19.811: ephone-1[0/10]:StationSoftKeyTemplateResMessage000343: May 5 16:13:19.819: ephone-1[0/10]:StationSoftKeySetReqMessage000344: May 5 16:13:19.819: ephone-1[0/10]:StationSoftKeySetResMessage000345: May 5 16:13:19.839: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatReqMessage from ephone line 3000346: May 5 16:13:19.839: ephone-1[0/10]:StationLineStatReqMessage ephone line 3 DN 8 = 2764 desc = Sonus label = 2764-3000347: May 5 16:13:19.839: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatResMessage sent to ephone (1 of 6)000348: May 5 16:13:19.847: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatReqMessage from ephone line 2000349: May 5 16:13:19.847: ephone-1[0/10]:StationLineStatReqMessage ephone line 2 DN 7 = 2764 desc = Sonus label = 2764-2000350: May 5 16:13:19.847: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatResMessage sent to ephone (2 of 6)000351: May 5 16:13:19.851: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatReqMessage from ephone line 1000352: May 5 16:13:19.851: ephone-1[0/10]:StationLineStatReqMessage ephone line 1 DN 1 = 2764 desc = Sonus label = 2764-1000353: May 5 16:13:19.851: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatResMessage sent to ephone (3 of 6)000354: May 5 16:13:19.879: ephone-1[0/10][SEP000DBCD80C50]:StationSpeedDialStatReqMessage speed 3000355: May 5 16:13:19.879: ephone-1[0/10][SEP000DBCD80C50]:No speed-dial set 3000356: May 5 16:13:19.879: ephone-1[0/10]:StationSpeedDialStatMessage sent61


000357: May 5 16:13:19.891: ephone-1[0/10][SEP000DBCD80C50]:StationSpeedDialStatReqMessage speed 2000358: May 5 16:13:19.891: ephone-1[0/10][SEP000DBCD80C50]:No speed-dial set 2000359: May 5 16:13:19.891: ephone-1[0/10]:StationSpeedDialStatMessage sent000360: May 5 16:13:19.895: ephone-1[0/10][SEP000DBCD80C50]:StationSpeedDialStatReqMessage speed 1000361: May 5 16:13:19.895: ephone-1[0/10][SEP000DBCD80C50]:Speed Dial 1: 103000362: May 5 16:13:19.895: ephone-1[0/10]:StationSpeedDialStatMessage sent000363: May 5 16:13:19.939: ephone-1[0/10][SEP000DBCD80C50]:Skinny Available Lines 3 set for socket [10]000364: May 5 16:13:19.939: ephone-1[0/10]:SkinnyCompleteRegistrationForAvailLines000365: May 5 16:22:13.451: %SYS-5-CONFIG_I: Configured from console by jets on console000366: May 5 17:53:33.027: ephone-1[0/10]:MediaPathEventMessageCME1#

c3825nm#Use the sh telephony-service tftp-bindings command to ensure that the specific IP phone configuration files are correct.

c3825nm#sh telephony-service tftp-bindingstftp-server system:/its/SEPDEFAULT.cnftftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnftftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xmltftp-server system:/its/ATADefault.cnf.xmltftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xmltftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xmltftp-server system:/its/united_states/7960-dictionary.xml alias English_United_States/7960-dictionary.xmltftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7960-kate.xmltftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_States/SCCP-dictionary.xmltftp-server system:/its/XMLDefault-1.cnf.xml alias SEP0013C4AAC149.cnf.xmltftp-server system:/its/XMLDefault7970.cnf.xml alias SEP000F23F9890D.cnf.xmltftp-server system:/its/XMLDefault7912.cnf.xml alias SEP000F8FE13E8C.cnf.xmltftp-server system:/its/XMLDefault7960.cnf.xml alias SEP001201D75FE2.cnf.xmltftp-server system:/its/XMLDefault7960.cnf.xml alias SEP001201F5CEBF.cnf.xmltftp-server system:/its/XMLDefault7960.cnf.xml alias SEP000D28D77476.cnf.xmltftp-server system:/its/XMLDefault7920.cnf.xml alias SEP000D288E6D72.cnf.xml

62


Use the show voice translation-rule number command to verify the configuration rule.

CME1#sh voice translation-rule 1Translation-rule tag: 1

Rule 1: Match pattern: 2764 Replace pattern: 7322162764 Match type: none Replace type: none Match plan: none Replace plan: none





CME1#

To test the transition rule and translation profile use the test voice translation-rule number input-test-string command. CME1#test voice translation-rule 1 2764Matched with rule 1Original number: 2764 Translated number: 7322162764Original number type: none Translated number type: noneOriginal number plan: none Translated number plan: none

CME1#

63


6 Appendix A: Acronyms

Acronym TranslationADSL Asymmetric Digital Subscriber LineAIM Advanced Integration Module AAS Autonomous SystemATM Asynchronous Transfer ModeAT&T VPN AT&T Virtual Private NetworkBC Committed Burst BE Excess Burst or Best EffortBGP Border Gateway ProtocolBH Bursty HighBL Bursty LowBOE Branch Office ExtensionCAS Channel Associated SignalingCBWFQ Class Based Weighted Fair QueuingCCG Customer Configuration GuideCCS Common Channel SignalingCDR Committed Data RateCEF Cisco Express ForwardingCER Customer Edge RouterCHAP Challenge Handshake Authentication ProtocolCIR Committed Information RateCLI Command Line InterfaceCM Communications ManagerCME Call Manager ExpressCOS Class of ServiceCPE Customer Premise EquipmentCPU Central Processing UnitCRC Cyclic Redundancy CheckCRTP Compress Real Time ProtocolCSU/DSU Channel Service Unit / Data Service UnitCUBE Cisco Unified Border ElementCUCM Cisco Unified Communications ManagerCUCME Cisco Unified Communications Manager ExpressCUE Cisco Unity ExpressDID Direct Inward DialDS Down StreamDSCP Differentiated Service Code PointDSL Digital Subscriber LineDSP Digital Signal ProcessorsDTMF Dual Tone Multi FrequencyE&M Ear & Mouth

64


EF Expedient ForwardingePVC Enhanced Permanent Virtual CircuitFR Frame RelayFXO Foreign Exchange OfficeFXS Foreign Exchange StationGSM FR Global System for Mobile communications Full RateHDV High Density VoiceHWIC High-speed WAN Interface CardIAR Inbound Alternate RoutingIETF Internet Engineering Task ForceIMA Inverse Multiplexing over ATM IOS Internetwork Operation SystemIP Internet ProtocolIPBE Internet Protocol Border ElementIPSEC Internet Protocol SecurityISR Integrated Services RouterITU-T International Telecommunication Union -

TelecommunicationsGW GatewayLAN Local Area NetworkLFI Link Fragmentation and InterleavingLLQ Low Latency QueuingLD Long DistanceMLPPP Multi-Link Point-to-Point ProtocolMM Multi MediaMOW Most Of WorldMTU Maximum Transmission UnitNAT Network Address TranslationNET Network Equipment TechnologiesNM Network ModuleNPE Network Processing EngineOAM Operation Administration & MaintenanceOCS Office Communication ServerPA Port AdapterPAT Port Address TranslationPBX Private Branch ExchangePC Personal ComputerPCR Peak Cell RatePER Provider Edge RouterPOS Packet over SONETPOTS Plain Old Telephone ServicePPP Point-to-Point ProtocolPQ Priority QueuePRI Primary Rate Interface

65


PSAP Public Safety Answering PointPSTN Public Switched Telephone NetworkPVC Permanent Virtual CircuitPVDM Packet Voice DSP ModuleQOS Quality of ServiceQSIG Q SignalingRC ReceiveRFC Request for CommentRT Real TimeRTCP Real Time Control ProtocolRTP Real Time ProtocolSBC Session Border ControllerSCCP Skinny Call Control ProtocolSCR Sustainable Cell RateSHDSL Single-Pair High-Speed Digital Subscriber LineSIP Session Initiation ProtocolSM Session ManagerSPE Synchronous Payload EnvelopeTAC Technical Assistance CenterTC Time IntervalTDM Time Division MultiplexingTN Telephone NumberTX TransmitUDP User Datagram ProtocolUS Up Stream or United StatesVAD Voice Activity DetectionVCI Virtual Circuit IdentifierVLAN Virtual Local Area NetworkVNI Voice Network InfrastructureVoIP Voice over Internet ProtocolVPI Virtual Path IdentifierVPN Virtual Private NetworkVQM Voice Quality MonitorVT Virtual TemplateWAN Wide Area NetworkWFQ Weighted Fair QueuingWIC WAN Interface Card

66


This Customer Configuration Guide ("CCG") is offered as a convenience to AT&T's customers. The specifications and information regarding the product in this CCG are subject to change without notice. All statements, information, and recommendations in this CCG are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this CCG.

In no event shall AT&T or its suppliers be liable for any indirect, special, consequential, or incidental damages, including, without limitation, lost profits or loss or damage arising out of the use or inability to use this CCG, even if AT&T or its suppliers have been advised of the possibility of such damage.

67

voavpn cucme 7.0(1) ccg - at&t › bvoip › wp-content › themes … · web viewip address...

Documents