VoAVPN CUCME 7.0(1) CCG - AT&T Business
IP Flexible Reach on AT&T VPN
Cisco Unified Communications Manager Express 7.0(1)
Customer Configuration Guide
Issue 1.0
3/7/2011
Table of Contents
1 Introduction 3
1.1 Overview 3
1.2 Cisco Unified Communications Manager Express Site 4
1.3 Features Supported 5
1.4 Features not supported 6
1.5 Platforms & Access Types 6
2 AT&T VPN WAN Access Configuration 6
2.1 T1 PPP 6
2.2 NxT1 MLPPP (2-8 T1s) 7
2.3 Traffic Classification and Queuing Techniques 10
2.4 Class of Service Configuration Example 12
2.5 Routing 13
2.5.1 Additional Instructions for the Voice Quality Monitor 13
2.5.2 Probe IP Addressing & Routing 14
2.5.3 TAP 16
3 Cisco Unified Communications Manager Express Configuration 16
3.1 Cisco Unified Communications Manager Express Version 16
3.2 DHCP Server 17
3.3 NTP Server 18
3.4 Install CME Application Files 18
3.5 CME Telephony Parameters 20
3.6 Dialing Restrictions 23
3.7 Phone Configuration 25
3.7.1 Ephone-dn 25
3.7.2 Ephone 26
3.8 Conference 27
3.9 VoIP Connectivty to AT&T IP Flexible Reach 31
3.10 Codec configuration 31
3.11 SIP Profile 32
3.12 Privacy with SIP (Optional) 32
3.13 Calling/Called Party Number Translation Configuration 33
3.14 Outbound Calls to AT&T 34
3.15 Inbound Calls from AT&T 36
4 Unity Express Configuration 36
5 Troubleshooting 39
6 Appendix A: Acronyms 64
Introduction
AT&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 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.
[pic]
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.
• 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 . 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.
3 Features Supported
• Basic Call using G.729 or G.711ulaw
• Calling Party Number Presentation
• 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
4 Features not supported
• Fax support on CME
5 Platforms & Access Types
Table 1: Cisco ISR Platform IP Telephony and Memory Specifications
|Router Platform |Max. IP Phones |Max. Ephone-dn |Minimum DRAM |Minimum Flash |
| | |Virtual Ports | | |
|Cisco 2811 |36 |144 |256 |64 |
|Cisco 2821 |48 |144 |256 |64 |
|Cisco 2851 |96 |288 |256 |64 |
|Cisco 3825 |168 |500 |256 |64 |
|Cisco 3845 |240 |720 |256 |64 |
Following are the access types that can be ordered:
|Access Type |Speed (bit/s) |Fragmentation |CRTP |
|T1 PPP access |1024K to 1536K |no |no |
|NXT1 MLPPP access |N = 2 to 8 T1s |no |no |
AT&T VPN WAN Access Configuration
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.
Example of T1 PPP Access on ISR:
interface Serial0/1/0
ip address
max-reserved-bandwidth 100
encapsulation ppp
service-policy output SHAPE_PPP
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 ”. 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 0
card 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
!
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
ip address
load-interval 30
ppp chap hostname ??
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
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
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
ppp multilink
ppp multilink group 1
!
interface Serial0/1/1:0
no ip address
encapsulation ppp
tx-ring-limit 2
tx-queue-limit 2
ppp chap hostname
ppp multilink
ppp multilink group 1
interface Loopback0
description "FlexReach assigned Media+Signaling Address"
ip address
!
interface Loopback1
description "FlexReach assigned VoIP Probe Address"
ip address
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 255.255.255.0
!
interface GigabitEthernet0/0.2
description Branch LAN
ip address 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
encapsulation ppp
load-interval 30
max-reserved-bandwidth 100
service-policy output SHAPE_PPP
ip nat outside
!
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
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.
[pic]
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.
4 Class of Service Configuration Example
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-Traffic
class-map match-any BGP
match access-group name BGP
class-map match-any COS5
match access-group name COS5-Traffic
class-map match-any COS4
match access-group name COS4-Traffic
class-map match-any COS3
match access-group name COS3-Traffic
class-map match-any COS2
match access-group name COS2-Traffic
match access-group name BGP
class-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 cs6
policy-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 default
policy-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
ip access-list extended COS2-Traffic
permit udp any any eq 2082
permit udp any eq 2082 any
ip access-list extended COS2V-Traffic
permit tcp any any range 3230 3231
permit udp any any range 3230 3235
ip access-list extended COS3-Traffic
permit udp any any eq 2083
permit udp any eq 2083 any
ip access-list extended COS5-Traffic
permit udp any any eq 110
permit udp any eq 110 any
ip access-list extended RTP
permit udp any range 16384 32767 any range 16384 32767
ip access-list extended SCCP
permit tcp any range 2000 2003 any
permit tcp any any range 2000 2003
ip access-list extended SIP
permit udp any eq 5060 any
permit udp any any eq 5060
!
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
!
1 Additional Instructions for the Voice Quality Monitor
Figure 3 shows the WineyeQ probe topology at the CUCME store location.
[pic]
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
Step 2) Create a new loopback interface, ethernet interface and the necessary network address translation (NAT) for VoIP probe communication.
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
ip address < IP address assigned by IP Flex for SIP Signaling> 255.255.255.255
int loopback
ip address < IP address assigned by IP Flex with lower fourth octet> 255.255.255.255
int
ip address 10.255.255.253 255.255.255.252
ip nat inside
int
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 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
network 32.252.97.1 mask 255.255.255.255
network 32.252.97.2 mask 255.255.255.255
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.
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.
1 Cisco Unified Communications Manager Express Version
Cisco IOS Version
CME1#show version
Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version 12.4(22)T5, RELEASE SOFTWARE (fc2)
Technical Support:
Copyright (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 minutes
System returned to ROM by reload at 11:55:29 EDT Mon Mar 29 2010
System restarted at 11:56:55 EDT Mon Mar 29 2010
System image file is "flash:c2800nm-adventerprisek9-mz.124-22.T5.bin"
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to 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:
If you require further assistance please contact us by sending email to
export@.
Cisco 2851 (revision 53.51) with 763904K/22528K bytes of memory.
Processor board ID FTX1343AJ6T
2 Gigabit Ethernet interfaces
2 Serial(sync/async) interfaces
1 terminal line
1 Virtual Private Network (VPN) Module
1 cisco Integrated Service Engine(s)
Cisco Unity Express 7.0.5 in slot 1
DRAM 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
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 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 Loopback0 IP Address
3 NTP Server
ntp master 2
ntp 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.
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-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 software
After 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 t 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 t flash:
After all required files are installed, use the “show flash” command to list the files installed on flash memory.
cme-124-22T2.zip
7970-backgrounds.tar
app-cme-did-2.0.0.0.tar
cme-bacd-2.1.2.2.tar
cmterm-3951-sip.8-1-1.tar
cmterm-7911_7906-sccp.8-3-3S.tar
CP7921G-1.1.1.TAR
cmterm-7931-sccp.8-3-3S.tar
cmterm-7941_7961-sccp.8-3-3S.tar
cmterm-7942_7962-sccp.8-3-3S.tar
cmterm-7945_7965-sccp.8-3-3S.tar
cmterm-7970_7971-sccp.8-3-3S.tar
cmterm-7975-sccp.8-3-3S.tar
P00308000500.tar
ringtone.tar
cmterm_7920.4.0-03-02.bin
cmterm_7936.3-3-15-0.bin
cmterm_7985.4-1-4-0.bin
CP7902080002SCCP060817A.sbin
CP7912080003SCCP070409A.sbin
music-on-hold.au
P00405000700.bin
P00405000700.sbn
P00503021500.bin
S00105000300.sbn
apps37sccp.1-1-1-1.bin
ATA030203SCCP051201A.zup
B015-1-0-2.SBN
B016-1-0-2.SBN
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
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 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. |
|• 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:
telephony-service
load 7905 P7905080001SCCP051117A!Select the firmware load file for the 7905 phone
load 7906 SCCP11.8-3-3S.loads! Select the firmware load file for the 7906 phone
load 7911 SCCP11.8-3-3S.loads! Select the firmware load file for the 7911 phone
load 7912 CP7912080003SCCP070409A.sbin! Select the firmware load file for the 7912 phone
load 7960-7940 P00308000500! Select the firmware load file for the 7960/7940 phone
load 7961 SCCP41.8-3-3S.loads! Select the firmware load file for the 7961 phone
load 7965 SCCP45.8-3-3S.loads! Select the firmware load file for the 7965 phone
load 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..
Example Configuration
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 16
sdspfarm tag 1 MTP0026cb5c16e1
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
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 name
2. Associating each COR custom name with CoR list name that will be applied to dial-peer associated with specific destination patterns
3. 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 name
name Local
name Longdistance
name 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)
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:
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
corlist incoming COS1! Apply the COR CoS COS1 list to the incoming calls from this phone
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.
1 Ephone-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
• 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
2 Ephone
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
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 5
ephone 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
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:
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 GigabitEthernet0/0.1
sccp 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 group
sccp ! 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.
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 mode
sdspfarm 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 added
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)
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
!
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.
Example Configuration
sccp local GigabitEthernet0/1
sccp ccm 10.30.10.1 identifier 2851 version 7.0 Customer LAN Address
sccp
!
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.
• 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 :
• 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.
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
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
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 "" ""
** 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...
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:
2. P-Asserted-Identity message header is added with privacy id field
P-Asserted-Identity: sip:7322162764@135.16.206.13
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"
Contact:
** Note – If this option is used all registered ephones will be set to privacy.
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 criteria
2. 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/
!
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.
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:
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:
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
session protocol sipv2
session target ipv4:
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:
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:
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:
dtmf-relay rtp-nte
dial-peer hunt 1
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.
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.... (assumes 10-digits being sent by Flex IPBE)
dtmf-relay rtp-nte
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 !
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
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 !!
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
!
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
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 Interfac
e 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 DSP
SIMM 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 PeerId
alias exec rtp show voip rtp con
alias exec codec show call active voice | in Cod
alias exec sccp show sccp conn
alias exec connect show voice call summ | in CON
sh ver
sh run
sh flash:
sh call app voice summ
If the calls still fail - would need the below debugs:
deb voip ivr script
deb vpm sig
deb voip dialpeer inout
deb voip ccapi inout
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 configuration
CME1#sh telephony-service all
CONFIG (Version=7.0(1))
=====================
Version 7.0(1)
Cisco Unified Communications Manager Express
For on-line documentation please see:
s_home.html
ip source-address 135.16.206.13 port 2000
no auto-reg-ephone
load 7905 CP7905080001SCCP051117A
load 7906 SCCP11.8-3-3S
load 7911 SCCP11.8-3-3S
load 7912 CP7912080003SCCP070409A.sbin
load 7960-7940 P00308000500
load 7961 SCCP41.8-3-3S
load 7965 SCCP45.8-3-3S
load 7975 SCCP75.8-3-3S
max-ephones 20
max-dn 40
max-conferences 8 gain 0
dspfarm units 1
dspfarm transcode sessions 16
dspfarm 1 MTP0026cb5c16e1
conference software
privacy
no privacy-on-hold
hunt-group report delay 1 hours
hunt-group logout DND
max-redirect 5
voicemail 7322162767
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
phone service rtcp 1
moh flash:/CME/music-on-hold.au
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
secondary-dialtone 9
transfer-pattern 1..........
transfer-pattern .T
keepalive 30 auxiliary 30
timeout interdigit 5
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
system message AT&T Store #70
web admin system name cmeadmin password jets
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp 7960 Mar 08 2010 15:36:00
transfer-system full-consult dss
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
ephone-dn 1 dual-line
number 2764
name Store #70
description Sonus
label 2764-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
cor incoming COS1
ephone-dn 2 dual-line
number 2765
name Store #70
description Sonus
label 2765-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 3
number 8000..........
preference 0 secondary 9
huntstop
call-waiting beep
mwi on
ephone-dn 4
number 8001..........
preference 0 secondary 9
huntstop
call-waiting beep
mwi off
ephone-dn 5 dual-line
number 7322162766
name Store #70
description Sonus
label 2766-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 6 dual-line
number 2162768
name Store #70
description Sonus
label 2768-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 7 dual-line
number 2764
name Store #76
description Sonus
label 2764-2
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 8 dual-line
number 2764
name Store #70
description Sonus
label 2764-3
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 9 dual-line
number 2766
name Store #70
description Sonus
label 2766-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
ephone-dn 10 dual-line
number 3764
name Store #70
description HIPCS/NSN
label 3764-1
preference 0 secondary 9
huntstop
no huntstop channel
call-waiting beep
ephone-dn 11 dual-line
number 3765
name Store #70
description HIPCS/NSN
label 3765-1
preference 0 secondary 9
huntstop
no huntstop channel
call-waiting beep
Number of Configured ephones 7 (Registered 6)
ephone 1
Device Security Mode: Non-Secure
mac-address 000D.BCD8.0C50
type 7960
button 1:1 2:7 3:8
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
ephone 2
Device Security Mode: Non-Secure
mac-address 001A.2FF5.8E69
type 7940
button 1:2
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference endcall
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 3
Device Security Mode: Non-Secure
mac-address 0019.567E.8C89
type 7961
button 1:6
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 4
Device Security Mode: Non-Secure
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 5
Device Security Mode: Non-Secure
mac-address 0013.C326.841F
type 7905
button 1:9
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 6
Device Security Mode: Non-Secure
mac-address 001F.CA34.B2BB
type 7965
button 1:10
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 7
Device Security Mode: Non-Secure
mac-address 0021.5553.8707
type 7975
button 1:11
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
voice-port 50/0/1
station-id number 2764
station-id name Store #70
timeout interdigit 5
timeout ringing 5
!
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 5
voice-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
!
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 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 20004 pots
destination-pattern 8001..........
huntstop
progress_ind setup enable 3
port 50/0/4
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
dial-peer voice 20006 pots
destination-pattern 2162768$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/6
dial-peer voice 20007 pots
destination-pattern 2764$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/7
dial-peer voice 20008 pots
destination-pattern 2764$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/8
dial-peer voice 20009 pots
destination-pattern 2766$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/9
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 50/0/11
tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_States/7920-font.xml
tftp-server system:/its/united_states/7960-dictionary.xml alias English_United_States/7960-dictionary.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7960-kate.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7920-kate.xml
tftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_States/SCCP-dictionary.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/f alias f
tftp-server system:/its/vrf1/f.xml alias f.xml
tftp-server system:/its/vrf1/f.xml alias f.xml
Gather the customer’s dial plan. Make sure that the customer’s private dial plan is properly provisioned on the gatekeeper.
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 ephone
Number of Configured ephones 7 (Registered 6)
ephone 1
Device Security Mode: Non-Secure
mac-address 000D.BCD8.0C50
type 7960
button 1:1 2:7 3:8
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 2
Device Security Mode: Non-Secure
mac-address 001A.2FF5.8E69
type 7940
button 1:2
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference endcall
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 3
Device Security Mode: Non-Secure
mac-address 0019.567E.8C89
type 7961
button 1:6
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 4
Device Security Mode: Non-Secure
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 5
Device Security Mode: Non-Secure
mac-address 0013.C326.841F
type 7905
button 1:9
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 6
Device Security Mode: Non-Secure
mac-address 001F.CA34.B2BB
type 7965
button 1:10
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
ephone 7
Device Security Mode: Non-Secure
mac-address 0021.5553.8707
type 7975
button 1:11
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-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 1
ephone 1
Device Security Mode: Non-Secure
mac-address 000D.BCD8.0C50
type 7960
button 1:1 2:7 3:8
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
CME1#sh telephony-ser ephone-dn | beg ephone-dn 1
ephone-dn 1 dual-line
number 2764
name Store #70
description Sonus
label 2764-1
preference 0 secondary 9
huntstop
no huntstop channel
call-forward busy 7322162767
call-forward noan 7322162767 timeout 5
call-waiting beep
cor 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 1
50/0/1 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
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 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 20004 pots
destination-pattern 8001..........
huntstop
progress_ind setup enable 3
port 50/0/4
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
dial-peer voice 20006 pots
destination-pattern 2162768$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/6
dial-peer voice 20007 pots
destination-pattern 2764$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/7
dial-peer voice 20008 pots
destination-pattern 2764$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/8
dial-peer voice 20009 pots
destination-pattern 2766$
huntstop
call-forward busy 7322162767
call-forward noan 7322162767
progress_ind setup enable 3
port 50/0/9
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 50/0/11
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-1
ephone-1[0] Mac:000D.BCD8.0C50 TCP socket:[3] activeLine:0 REGISTERED in SCCP ve
r 11/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8
IP:10.30.10.82 50127 Telecaster 7960 keepalive 14454 max_line 6 available_line3
button 1: dn 1 number 2764 CH1 IDLE CH2 IDLE
button 2: dn 7 number 2764 CH1 IDLE CH2 IDLE
button 3: dn 8 number 2764 CH1 IDLE CH2 IDLE
speed dial 1:103 Speed-dial
Preferred Codec: g711ulaw
Identify the mac address of the phone using command:
CME1#sh telephony-service ephone | beg ephone 1
ephone 1
Device Security Mode: Non-Secure
mac-address 000D.BCD8.0C50
type 7960
button 1:1 2:7 3:8
speed-dial 1 103 label Speed-dial
keepalive 30 auxiliary 30
multicast-moh
max-calls-per-button 8
busy-trigger-per-button 0
Always send media packets to this router: No
Preferred codec: g711ulaw
keep-conference
conference drop-mode never
conference add-mode all
conference admin: No
privacy: Yes
privacy button: No
user-locale US
network-locale US
!
Turn on debug ephone reg for the specific ephone –
CME1#debug ephone register mac-address 000D.BCD8.0C50
EPHONE registration debugging is enabled for phone 000D.BCD8.0C50
Clear the log
CME1#clear logg
Clear logging buffer [confirm]y
CME1#
Reset the phone –
CME1#config t
Enter configuration commands, one per line. End with CNTL/Z.
CME1(config)#ephone 1
CME1(config-ephone)#reset
reseting 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 sent
000290: May 5 16:09:46.471: ephone-1[0/3][SEP000DBCD80C50]:Phone Unregistered on socket [3] SEP000DBCD80C50
000291: 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.
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 sent
000290: May 5 16:09:46.471: ephone-1[0/3][SEP000DBCD80C50]:Phone Unregistered on socket [3] SEP000DBCD80C50
000291: 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 10.30.10.82
000295: May 5 16:13:19.587: skinny_add_socket 2 10.30.10.82 50095
000296: May 5 16:13:19.595: %IPPHONE-6-REG_ALARM: 25: Name=SEP000DBCD80C50 Load=8.1(1.0) Last=Initialized
000297: May 5 16:13:19.595: ephone-(1)[10] StationRegisterMessage (5/6/21) from 10.30.10.82
000298: May 5 16:13:19.595: ephone-(1)[10] Register StationIdentifier DeviceName SEP000DBCD80C50
000299: May 5 16:13:19.595: ephone-(1)[10] StationIdentifier Instance 1 deviceType 7
000300: May 5 16:13:19.595: ephone-1[0/3]:stationIpAddr 10.30.10.82
000301: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:maxStreams 0
000302: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:From Phone raw protocol Ver 0x8560000B
000303: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:protocol Ver 0x8560000B
000304: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:phone-size 57240 dn-size 784
000305: May 5 16:13:19.595: ephone-(1) Allow any Skinny Server IP address 135.16.206.13
000306: May 5 16:13:19.595: ephone-1[0/3][SEP000DBCD80C50]:Found entry 0 for 000DBCD80C50
000307: May 5 16:13:19.599: ephone-1[0/3][SEP000DBCD80C50]:socket change 3 to 10
000308: May 5 16:13:19.599: ephone-1[0/3][SEP000DBCD80C50]:DisAssociate: Closed socket 3 for unregistered phone
000309: May 5 16:13:19.599: ephone-1[0/-1][SEP000DBCD80C50]:FAILED: CLOSED old socket -1
000310: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:***Force device subtype to 0
000311: 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 10
000314: 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 50095
000316: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:Signal protocol ver 9 to phone with ver 11
000317: May 5 16:13:19.599: ephone-1[0/10][SEP000DBCD80C50]:Date Format M/D/Y
000318: May 5 16:13:19.599: ephone-1[0/10]:RegisterAck sent to sockettype ephone socket 10: keepalive period 30 use sccp-version 9
000319: May 5 16:13:19.599: ephone-1[0/10]:CapabilitiesReq sent
000320: May 5 16:13:19.607: ephone-1[0/10]:HeadsetStatusMessage
000321: May 5 16:13:19.639: ephone-1[0/10]:MediaPathCapabilitiesMessage
000322: May 5 16:13:19.643: ephone-1[0/10]:MediaPathEventMessage
000323: May 5 16:13:19.643: ephone-1[0/10]:MediaPathCapabilitiesMessage
000324: May 5 16:13:19.647: ephone-1[0/10]:MediaPathEventMessage
000325: May 5 16:13:19.647: ephone-1[0/10]:CapabilitiesRes received
000326: May 5 16:13:19.647: ephone-1[0/10][SEP000DBCD80C50]:Caps list 8
WideBand_256K 120 ms, is_mtp 0
G711Ulaw64k 40 ms, is_mtp 0
G711Alaw64k 40 ms, is_mtp 0
G729AnnexB 60 ms, is_mtp 0
G729AnnexAwAnnexB 60 ms, is_mtp 0
G729 60 ms, is_mtp 0
G729AnnexA 60 ms, is_mtp 0
Unrecognized 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 10.30.10.82
000329: May 5 16:13:19.659: skinny_add_socket 2 10.30.10.82 50096
000330: May 5 16:13:19.659: ephone-1[0/10]:MediaPathEventMessage
000331: May 5 16:13:19.807: ephone-1[0/10]:MediaPathEventMessage
000332: May 5 16:13:19.807: ephone-1[0/10]:ButtonTemplateReqMessage
000333: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:StationButtonTemplateReqMessage set max presentation to 6
000334: May 5 16:13:19.807: ephone-1[0/10]:CheckAutoReg
000335: May 5 16:13:19.807: ephone-1[0/10]:AutoReg is disabled
000336: 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
000339: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:ButtonTemplate buttonCount=6 totalButtonCount=6 buttonOffset=0
000340: May 5 16:13:19.807: ephone-1[0/10][SEP000DBCD80C50]:Configured 3 speed dial buttons
000341: May 5 16:13:19.811: ephone-1[0/10]:StationSoftKeyTemplateReqMessage
000342: May 5 16:13:19.811: ephone-1[0/10]:StationSoftKeyTemplateResMessage
000343: May 5 16:13:19.819: ephone-1[0/10]:StationSoftKeySetReqMessage
000344: May 5 16:13:19.819: ephone-1[0/10]:StationSoftKeySetResMessage
000345: May 5 16:13:19.839: ephone-1[0/10][SEP000DBCD80C50]:StationLineStatReqMessage from ephone line 3
000346: May 5 16:13:19.839: ephone-1[0/10]:StationLineStatReqMessage ephone line 3 DN 8 = 2764 desc = Sonus label = 2764-3
000347: 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 2
000349: May 5 16:13:19.847: ephone-1[0/10]:StationLineStatReqMessage ephone line 2 DN 7 = 2764 desc = Sonus label = 2764-2
000350: 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 1
000352: May 5 16:13:19.851: ephone-1[0/10]:StationLineStatReqMessage ephone line 1 DN 1 = 2764 desc = Sonus label = 2764-1
000353: 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 3
000355: May 5 16:13:19.879: ephone-1[0/10][SEP000DBCD80C50]:No speed-dial set 3
000356: May 5 16:13:19.879: ephone-1[0/10]:StationSpeedDialStatMessage sent
000357: May 5 16:13:19.891: ephone-1[0/10][SEP000DBCD80C50]:StationSpeedDialStatReqMessage speed 2
000358: May 5 16:13:19.891: ephone-1[0/10][SEP000DBCD80C50]:No speed-dial set 2
000359: May 5 16:13:19.891: ephone-1[0/10]:StationSpeedDialStatMessage sent
000360: May 5 16:13:19.895: ephone-1[0/10][SEP000DBCD80C50]:StationSpeedDialStatReqMessage speed 1
000361: May 5 16:13:19.895: ephone-1[0/10][SEP000DBCD80C50]:Speed Dial 1: 103
000362: May 5 16:13:19.895: ephone-1[0/10]:StationSpeedDialStatMessage sent
000363: 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]:SkinnyCompleteRegistrationForAvailLines
000365: May 5 16:22:13.451: %SYS-5-CONFIG_I: Configured from console by jets on console
000366: May 5 17:53:33.027: ephone-1[0/10]:MediaPathEventMessage
CME1#
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-bindings
tftp-server system:/its/f
tftp-server system:/its/f alias f
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml
tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_States/7960-font.xml
tftp-server system:/its/united_states/7960-dictionary.xml alias English_United_States/7960-dictionary.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_States/7960-kate.xml
tftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_States/SCCP-dictionary.xml
tftp-server system:/its/XMLDefault-f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
tftp-server system:/its/f.xml alias f.xml
Use the show voice translation-rule number command to verify the configuration rule.
CME1#sh voice translation-rule 1
Translation-rule tag: 1
Rule 1:
Match pattern: 2764
Replace pattern: 7322162764
Match type: none Replace type: none
Match plan: none Replace plan: none
Rule 2:
Match pattern: 2765
Replace pattern: 7322162765
Match type: none Replace type: none
Match plan: none Replace plan: none
Rule 3:
Match pattern: 2766
Replace pattern: 7322162766
Match type: none Replace type: none
Match plan: none Replace plan: none
Rule 4:
Match pattern: 3764
Replace pattern: 3143323764
Match type: none Replace type: none
Match plan: none Replace plan: none
Rule 5:
Match pattern: 3765
Replace pattern: 3143323765
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 2764
Matched with rule 1
Original number: 2764 Translated number: 7322162764
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
CME1#
Appendix A: Acronyms
|Acronym |Translation |
|ADSL |Asymmetric Digital Subscriber Line |
|AIM |Advanced Integration Module A |
|AS |Autonomous System |
|ATM |Asynchronous Transfer Mode |
|AT&T VPN |AT&T Virtual Private Network |
|BC |Committed Burst |
|BE |Excess Burst or Best Effort |
|BGP |Border Gateway Protocol |
|BH |Bursty High |
|BL |Bursty Low |
|BOE |Branch Office Extension |
|CAS |Channel Associated Signaling |
|CBWFQ |Class Based Weighted Fair Queuing |
|CCG |Customer Configuration Guide |
|CCS |Common Channel Signaling |
|CDR |Committed Data Rate |
|CEF |Cisco Express Forwarding |
|CER |Customer Edge Router |
|CHAP |Challenge Handshake Authentication Protocol |
|CIR |Committed Information Rate |
|CLI |Command Line Interface |
|CM |Communications Manager |
|CME |Call Manager Express |
|COS |Class of Service |
|CPE |Customer Premise Equipment |
|CPU |Central Processing Unit |
|CRC |Cyclic Redundancy Check |
|CRTP |Compress Real Time Protocol |
|CSU/DSU |Channel Service Unit / Data Service Unit |
|CUBE |Cisco Unified Border Element |
|CUCM |Cisco Unified Communications Manager |
|CUCME |Cisco Unified Communications Manager Express |
|CUE |Cisco Unity Express |
|DID |Direct Inward Dial |
|DS |Down Stream |
|DSCP |Differentiated Service Code Point |
|DSL |Digital Subscriber Line |
|DSP |Digital Signal Processors |
|DTMF |Dual Tone Multi Frequency |
|E&M |Ear & Mouth |
|EF |Expedient Forwarding |
|ePVC |Enhanced Permanent Virtual Circuit |
|FR |Frame Relay |
|FXO |Foreign Exchange Office |
|FXS |Foreign Exchange Station |
|GSM FR |Global System for Mobile communications Full Rate |
|HDV |High Density Voice |
|HWIC |High-speed WAN Interface Card |
|IAR |Inbound Alternate Routing |
|IETF |Internet Engineering Task Force |
|IMA |Inverse Multiplexing over ATM |
|IOS |Internetwork Operation System |
|IP |Internet Protocol |
|IPBE |Internet Protocol Border Element |
|IPSEC |Internet Protocol Security |
|ISR |Integrated Services Router |
|ITU-T |International Telecommunication Union - Telecommunications |
|GW |Gateway |
|LAN |Local Area Network |
|LFI |Link Fragmentation and Interleaving |
|LLQ |Low Latency Queuing |
|LD |Long Distance |
|MLPPP |Multi-Link Point-to-Point Protocol |
|MM |Multi Media |
|MOW |Most Of World |
|MTU |Maximum Transmission Unit |
|NAT |Network Address Translation |
|NET |Network Equipment Technologies |
|NM |Network Module |
|NPE |Network Processing Engine |
|OAM |Operation Administration & Maintenance |
|OCS |Office Communication Server |
|PA |Port Adapter |
|PAT |Port Address Translation |
|PBX |Private Branch Exchange |
|PC |Personal Computer |
|PCR |Peak Cell Rate |
|PER |Provider Edge Router |
|POS |Packet over SONET |
|POTS |Plain Old Telephone Service |
|PPP |Point-to-Point Protocol |
|PQ |Priority Queue |
|PRI |Primary Rate Interface |
|PSAP |Public Safety Answering Point |
|PSTN |Public Switched Telephone Network |
|PVC |Permanent Virtual Circuit |
|PVDM |Packet Voice DSP Module |
|QOS |Quality of Service |
|QSIG |Q Signaling |
|RC |Receive |
|RFC |Request for Comment |
|RT |Real Time |
|RTCP |Real Time Control Protocol |
|RTP |Real Time Protocol |
|SBC |Session Border Controller |
|SCCP |Skinny Call Control Protocol |
|SCR |Sustainable Cell Rate |
|SHDSL |Single-Pair High-Speed Digital Subscriber Line |
|SIP |Session Initiation Protocol |
|SM |Session Manager |
|SPE |Synchronous Payload Envelope |
|TAC |Technical Assistance Center |
|TC |Time Interval |
|TDM |Time Division Multiplexing |
|TN |Telephone Number |
|TX |Transmit |
|UDP |User Datagram Protocol |
|US |Up Stream or United States |
|VAD |Voice Activity Detection |
|VCI |Virtual Circuit Identifier |
|VLAN |Virtual Local Area Network |
|VNI |Voice Network Infrastructure |
|VoIP |Voice over Internet Protocol |
|VPI |Virtual Path Identifier |
|VPN |Virtual Private Network |
|VQM |Voice Quality Monitor |
|VT |Virtual Template |
|WAN |Wide Area Network |
|WFQ |Weighted Fair Queuing |
|WIC |WAN Interface Card |
| | |
This Customer Configuration Guide ("CCG") is offered as a convenience to AT&T's customers. The specifications and information regarding the product in this CCG are subject to change without notice. All statements, information, and recommendations in this CCG are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this CCG.
In no event shall AT&T or its suppliers be liable for any indirect, special, consequential, or incidental damages, including, without limitation, lost profits or loss or damage arising out of the use or inability to use this CCG, even if AT&T or its suppliers have been advised of the possibility of such damage.
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