Cisco SRS Telephony Configuration

This chapter explains the required and optional tasks for configuring Cisco SRS Telephony Version 2.02.

Prerequisites for SRS Telephony Configuration

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The SRS Telephony router must be configured as the default router for the Cisco IP phones.

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Appropriate Cisco IP phone firmware versions must support the Cisco IP Phone 7960, Cisco IP Phone 7940, and Cisco IP Phone 7910 models: P003E302, P004E302, or higher. To get the appropriate Cisco IP phone firmware versions, go to the following URL: http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.

Note

You must purchase a feature license to turn on this new feature. You also need an account on Cisco.com to access the Cisco IP phone load versions.

•

Memory requirements are dependent on the platform and the number of supported Cisco IP phones. See the "Memory Requirements" sectionfor details.

Restrictions for Cisco SRS Telephony V2.02

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Does not support first-generation Cisco IP phones, such as Cisco IP Phone 30 VIP and Cisco IP Phone 12 SP+.

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Does not support other Cisco CallManager applications or services: Cisco IP SoftPhone, Cisco uOne—Voice and Unified Messaging Application, or Cisco IP Contact Center.

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Does not support any more directory numbers than the maximum specified number in the "Memory Requirements" section.

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Does not support Centralized Automatic Message Accounting (CAMA) trunks on Cisco 3660 routers.

Note

If you are in a state in the United States of America where there is a regulatory requirement for CAMA trunks to interface to 911 emergency services, and you want to connect more than 48 Cisco IP phones to the Cisco 3660 multiservice routers in your network, please contact your local Cisco account team for help in understanding and meeting the CAMA regulatory requirements.

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Voice over IP (VoIP) H.323, Voice over Frame Relay (VoFR), and Voice over ATM (VoATM) between Cisco gateways running Cisco IOS Release 12.2(11)T and using the H.323 nonstandard information element

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Foreign Exchange Office (FXO) and Foreign Exchange Station (FXS) loop-start (analog)

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T1 Channel Associated Signaling (CAS) with FXO and FXS ground-start signaling

•

The following Cisco IP phone function keys are not supported during SRS Telephony operation:

Information About Cisco SRS Telephony V2.02

The SRS Telephony feature provides Cisco CallManager with fallback support for Cisco IP phones attached to a Cisco router on your local network. The SRS Telephony feature enables routers to provide call-handling support for Cisco IP phones when they lose connection to remote primary, secondary, or tertiary Cisco CallManager installations, or when the WAN connection is down.

Cisco CallManager version 3.2 supports Cisco IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to the SRS Telephony feature, when the WAN connection between a router and Cisco CallManager failed, or connectivity with Cisco CallManager was lost for some reason, Cisco IP phones on the network became unusable for the duration of the failure. The SRS Telephony feature overcomes this problem and ensures that the Cisco IP phones offer continuous (yet, minimal) service by providing call-handling support for Cisco IP phones directly from the SRS Telephony router. The system automatically detects a failure and uses Simple Subnetwork Auto Provisioning (SNAP) technology to autoconfigure the branch office router to provide call processing for Cisco IP phones registered with the router. When the WAN link or connection to the primary Cisco CallManager is restored, call-handling reverts back to the primary Cisco CallManager.

Fallback Behavior

When Cisco IP phones lose contact with primary, secondary, and tertiary Cisco CallManagers, they must establish a connection to a local SRS Telephony router in order to ensure call-processing capability necessary to place and receive calls. The Cisco IP phone retains the IP address of the local SRS Telephony router as a default router in the Network Configuration area of the Settings menu. This list supports a maximum of five default router entries; however, Cisco CallManager accommodates a maximum of three entries. When a secondary Cisco CallManager is not available on the network, the local SRS Telephony router's IP address is retained as the standby connection for Cisco CallManager during normal operation.

When the WAN link fails, calls in progress are sustained for the duration of the call. Calls in transition and calls that have not yet connected are dropped and must be reinitiated once Cisco IP phones reestablish connection to their local SRS Telephony router. Telephone service remains unavailable from the time connection to the remote Cisco CallManager is lost until the Cisco IP phone establishes connection to the SRS Telephony router.

Note CallManager fallback mode telephone service is available only to those Cisco IP phones that are supported by an SRS Telephony router. Other Cisco IP phones on the network remain out of service until they are able to reestablish a connection with their primary, secondary, or tertiary Cisco CallManager.

The time taken to reestablish connection to a remote Cisco CallManager depends in part on the keepalive period set by Cisco CallManager itself. Typically, three times the keepalive period is required for a phone to discover that its connection to Cisco CallManager has failed. The default keepalive period is 30 seconds. If the phone has an active standby connection established with an SRS Telephony router, the fallback process takes 10 to 20 seconds after connection with Cisco CallManager is lost. An active standby connection to an SRS Telephony router exists only if the phone has the location of a single Cisco CallManager in its CallManager list. Otherwise, the phone activates a standby connection to its secondary Cisco CallManager.

If a Cisco IP phone has multiple Cisco CallManagers in its CallManager list, it progresses through its list of secondary and tertiary Cisco CallManagers before attempting to connect with its local SRS Telephony router. Therefore, the time that passes before the Cisco IP phone eventually establishes a connection with the SRS Telephony router increases with each attempt to contact a Cisco CallManager. Assuming that each attempt to connect to Cisco CallManager takes around 1 minute, the Cisco IP phone in question could remain offline for 3 minutes or more following a WAN link failure.

Note During a WAN connection failure, when SRS Telephony is enabled, Cisco IP phones display a message informing you that they are operating in Cisco CallManager fallback mode. The Cisco IP Phone 7960 and Cisco IP Phone 7940 display a "CM Fallback Service Operating" message and the Cisco IP Phone 7910 displays a "CM Fallback Service" message when operating in Cisco CallManager fallback mode. When the Cisco CallManager is restored, the message goes away and full Cisco IP phone functionality is restored.

While in CallManager fallback mode, Cisco IP phones periodically attempt to reestablish a connection with Cisco CallManager at the central office. When a connection is reestablished with Cisco CallManager, Cisco IP phones automatically cancel their registration with the SRS Telephony router. A Cisco IP phone cannot reestablish a connection with the primary Cisco CallManager at the central office if it is currently engaged in an active call.

Figure 2-1 shows a branch office with several Cisco IP phones connected to an SRS Telephony router. The router features connections to both a WAN link and the public switched telephone network (PSTN). The Cisco IP phones connect to their primary Cisco CallManager at the central office via this WAN link.

Figure 2-1 Branch Office Cisco IP Phones Connected to a Remote Central Cisco CallManager

Figure 2-2 shows the same branch office telephone network with the WAN connection down. In this situation, the Cisco IP phones use the SRS Telephony router as a fallback for their primary Cisco CallManager. The branch office Cisco IP phones are connected to the PSTN through the SRS Telephony router and are able to make and receive "off-net" calls.

How to Configure Cisco SRS Telephony

Configuring SRS Telephony on Routers to Support IP Phone Functions

Tip

When the SRS Telephony feature is enabled, Cisco IP phones do not need to be reconfigured while in Cisco CallManager fallback mode because phones retain the same configuration that was used with Cisco CallManager.

To configure SRS Telephony on the routers to support the Cisco IP phone functions, use the following commands beginning in global configuration mode:

SUMMARY STEPS

ip source-address ip-address [port port] [any-match | strict-match]

DETAILED STEPS


	Command	Purpose
Step 1	call-manager-fallback Example: Router(config)#	Enables SRS Telephony feature support and enters Cisco CallManager fallback mode.
Step 2	ip source-address ip-address [port port] [any-match \| strict-match] Example: Router(config-cm-fallback)# ip source-address 10.6.21.4 port 2002 strict-match	Enables the router to receive messages from the Cisco IP phones through the specified IP addresses and provides for strict IP address verification. The default port number is 2000.
Step 3	max-dn max-directory-numbers Example: Router(config-cm-fallback)# max-dn 12	Sets the maximum number of directory numbers or virtual voice ports that can be supported by the router. The default is 0. The maximum number is platform dependent. See the "Memory Requirements" section for further details. Note You must reboot the router in order to reduce the limit of the directory numbers or virtual voice ports after the maximum allowable number is configured.
Step 4	max-ephones max-phones Example: `Router(config-cm-fallback)# max-ephones 24`	Configures the maximum number of Cisco IP phones that can be supported by the router. The default is 0. The maximum number is platform dependent. See "Memory Requirements" section for further details. Note You must reboot the router in order to reduce the limit of the directory numbers or virtual voice ports after the maximum allowable number is configured.
Step 5	limit-dn {7910 \| 7940 \| 7960} max-lines Example: `Router(config-cm-fallback)# limit-dn 7910 2`	Limits the directory number lines on Cisco IP phones during CallManager fallback mode. Note You must configure this command during initial SRS Telephony router configuration, before any phone actually registers with the SRS Telephony router. However, you can modify the number of lines at a later time. The setting for maximum lines is from 1 to 6. The default number of maximum directory lines is set to 6. If there is any active phone with last line number greater than this limit, warning information is displayed for phone reset.

Verifying Cisco SRS Telephony

To verify that the SRS Telephony feature is enabled, perform the following steps:

Step 2

Enter the show call-manager-fallback all command to verify that SRS Telephony feature is enabled.

Step 3

Use the Settings display on the Cisco IP phones in your network to verify that the default router IP address on the phones matches the IP address of the SRS Telephony router.

Step 4

To temporarily block the TCP port 2000 Skinny Client Control Protocol (SCCP) connection for one of the Cisco IP phones to force the Cisco IP phone to lose its connection to the Cisco CallManager and register with the SRS Telephony router, perform the following steps:

Use the appropriate IP access-list command to temporarily disconnect a Cisco IP phone from the Cisco CallManager.

During a WAN connection failure, when SRS Telephony is enabled, Cisco IP phones display a message informing you that they are operating in Cisco CallManager fallback mode. The Cisco IP Phone 7960 and Cisco IP Phone 7940 display a "CM Fallback Service Operating" message and the Cisco IP Phone 7910 displays a "CM Fallback Service" message when operating in Cisco CallManager fallback mode. When the Cisco CallManager is restored, the message goes away and full Cisco IP phone functionality is restored.

Enter the no form of the appropriate access-list command to restore normal service for the phone.

Use the debug ephone register command to observe the registration process of the Cisco IP phone on the SRS Telephony router.

Use the show ephone command to display the Cisco IP phones that have registered to the SRS Telephony router.

Troubleshooting Tips

Step 1

To set keepalive debugging for the Cisco IP phone, use the debug ephone keepalive command.

Step 2

To set registration debugging for the Cisco IP phone, use the debug ephone register command.

Step 3

To set state debugging for the Cisco IP phone, use the debug ephone state command.

What to Do Next

Configuring Cisco SRS Telephony Optional Settings

Although following baseline Cisco SRS Telephony settings are not required, they are worth considering for possible configuration of the following features:

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Class of restriction (COR) on the dial peers associated with directory numbers

SUMMARY STEPS

alias tag number-pattern to alternate-number preference preference-value

dialplan-pattern tag prefix-pattern extension-length length [no-reg]

access-code {{fxo | e&m} dial-string | {bri | pri} dial-string} [direct-inward-dial]

11.

cor {incoming | outgoing} cor-list-name {cor-list-number starting-number - ending-number | default}

DETAILED STEPS


	Command	Purpose
Step 1	call-manager-fallback Example: Router(config)#	Enables SRS Telephony feature support and enters CallManager fallback mode.
Step 2	alias tag number-pattern to alternate-number preference preference-value Example: `Router(config-cm-fallback)# alias 1 60.. to 5001 preference 2`	(Optional) Allows routing of unmatched call destination to specific extension numbers with an associated dial-peer preference parameter.
Step 3	`date-format {mm-dd-yy \| dd-mm-yy}` Example: `Router(config-cm-fallback)# date-format dd-mm-yy`	(Optional) Sets the date display format on all Cisco IP phones registered with the router. The default is mm-dd-yy.
Step 4	time-format {12 \| 24} Example: `Router(config-cm-fallback)# time-format 24`	(Optional) Sets the time display format on all Cisco IP phones registered with the router. The default is 12 hours.
Step 5	keepalive seconds Example: Router(config-cm-fallback)# keepalive 60	(Optional) Configures the time interval between Cisco IP phones keepalive messages sent to the router when SRS Telephony is enabled. The range is from 10 to 65535 seconds. The default timeout is 30 seconds.
Step 6	default-destination telephone number Example: `Router(config-cm-fallback)# default-destination 40802`	(Optional) Assigns the default destination number for incoming telephone calls.
Step 7	dialplan-pattern tag prefix-pattern extension-length length [no-reg] Example: `Router(config-cm-fallback)# dialplan-pattern 1 40855550.. extension-length 4 no-reg`	(Optional) Creates a global prefix that can be used to expand the abbreviated extension numbers into fully qualified E.164 numbers. The extension-length keyword enables the system to convert a full E.164 telephone number back to an extension number for the purposes of caller-ID display, received, and missed call lists. The no-reg keyword provides dialing flexibility and prevents the E.164 numbers in the dial peer from registering to the gatekeeper. You have the option not to register some specific numbers to the gatekeeper so that those numbers can be used for other telephony services.
Step 8	transfer-pattern transfer-pattern Example: `Router(config-cm-fallback)# transfer-pattern 52540..`	(Optional) Allows telephone call transfer from Cisco IP phones to other phone numbers both within the local IP network and outside of the local IP network.
Step 9	`access-code {{fxo \| e&m} dial-string \| {bri \| pri} dial-string} [direct-inward-dial]` Example: `Router(config-cm-fallback)# access-code e&m 8`	(Optional) Configures trunk access codes for each type of line—Basic Rate Interface (BRI), E&M, Foreign Exchange Office (FXO), and Primary Rate Interface (PRI)—so that the Cisco IP phones can access the trunk lines while in Cisco CallManager fallback mode. The dial-string argument is used to set up temporary dial peers for each specified line type. The direct-inward-dial keyword enables you to set Direct Inward Dialing (DID) access for PRI and BRI trunk lines. Note The access-code command creates temporary dial peers in Cisco CallManager fallback mode. In many cases, you may already have the local PSTN ports configured with appropriate access codes provided by dial peers (for example, dial 9 to select an FXO PSTN line), in which case this command is not needed.
Step 10	voicemail phone-number Example: `Router(config-cm-fallback)# voicemail 914085551000`	(Optional) Configures the telephone number that is dialed when the message button on a Cisco IP phone is pressed.
Step 11	cor {incoming \| outgoing} cor-list-name {cor-list-number starting-number - ending-number \| default} Example: `Router(config-cm-fallback)# cor outgoing LockforPhoneC 1 5010 - 5020`	(Optional) Configures a COR on the dial peers associated with directory numbers. COR is used to specify which incoming dial peer can use which outgoing dial peer to make a call. Each dial peer can be provisioned with an incoming and an outgoing COR list. The cor-list-number argument ranges from 1 to 6. The starting-number - ending-number argument provides a range of directory numbers. The default keyword instructs the router to use an existing default COR list.
Step 12	`call-forward busy directory-number` Example: `Router(config-cm-fallback)# call-forward busy 50..`	(Optional) Configures call forwarding to another number when the Cisco IP phone is busy. Note The E.164 number you enter can contain one or more "." wildcard characters. The wildcard characters correspond to the right-justified digits in the directory number extension.
Step 13	`call-forward noan directory-number timeout seconds` Example: `Router(config-cm-fallback)# call-forward noan 5005 timeout 10`	(Optional) Configures call forwarding to another number when no answer is received from the Cisco IP phone. The timeout keyword sets the waiting time, in seconds, before the call is forwarded to another phone. The seconds range is from 3 to 6000 seconds. Note The E.164 number you enter can contain one or more "." wildcard characters. The wildcard characters correspond to the right-justified digits in the directory number extension.
Step 14	translate {called \| calling} translation-rule-tag Example: `Router(config-cm-fallback) translate called 20`	(Optional) Applies a translation rule to numbers dialed by Cisco IP phone users. The called keyword applies the translation rule to the outbound called party number. The calling keyword applies the translation rule to the inbound called party number. The translation-rule-tag argument is the reference number of the translation rule. Valid entries are from 1 to 2147483647. For further details, refer to the "Configuration Dial Plans, Dial Peers, and Digit Manipulation" chapter of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.
Step 15	timeouts interdigit seconds Example: `Router(config-cm-fallback)# timeouts interdigit 5`	(Optional) Configures the interdigit timeout value for all Cisco IP phones attached to the router. The interdigit timeout specifies the number of seconds that the system waits after the caller has entered the initial digit or a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. Note This value setting is important when using variable-length dial peer destination patterns (dial plans). For more information on setting dial plans, see the "Configuration Dial Plans, Dial Peers, and Digit Manipulation" chapter of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2. The seconds argument is the interdigit timeout wait time in seconds. A valid entry is an integer from 2 to 120 seconds. The default is 10 seconds.
Step 16	moh filename Example: `Router(config-cm-fallback)# moh minuet.wav`	(Optional) Configures music on hold. This feature supports .au and .wav format music files. Music on hold works only for G.711 calls and on-net VoIP and PSTN calls. For all other calls, callers hear a tone. Internal calls between Cisco IP phones do not get music on hold, instead, callers hear a tone. Note The music on hold file can be in .wav or .au file format; however, the file format must contain 8-bit 8 kHz data, for example, Consultative Committee for International Telegraph and Telephone (CCITT) a-law or u-law data format.

Troubleshooting Tips

•

To set detail debugging for the Cisco IP phones, use the debug ephone detail command.

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To set error debugging for the Cisco IP phones, use the debug ephone error command.

•

To set call statistics debugging for the Cisco IP phones, use the debug ephone statistics command.

•

To provide voice packet level debugging and print the contents of one voice packet in every 1024 voice packets, use the debug ephone pak command.

•

To provide raw low-level protocol debugging display for all SCCP messages, use the debug ephone raw command.

What to Do Next

Configuring Cisco SRS Telephony for Unity Voice-Mail Integration

For dual tone multifrequency (DTMF) integrations, information on how to route incoming or forwarded calls is sent by the telephone system in the form of DTMF digits. The DTMF digits are in the form of a pattern and depend on the voice-mail system connected to the Cisco SRS Telephony router. These patterns are required for the DTMF integration with most voice-mail systems. The DTMF integration configuration on the Cisco SRS Telephony router works with any analog voice-mail system. Voice-mail systems are designed to respond to DTMF after the system has answered the incoming calls. The tasks described in the following sections are required:

Note FXO hairpin forwarded calls to voice mail must have disconnect supervision from the central office.

Configuring DTMF Patterns on the Router

The Cisco SRS Telephony router provides flexibility for the integration with any legacy voice-mail system. You can configure multiple tags and tokens for each pattern, depending on the voice-mail system and type of access. The tag in the configuration pattern must match the number defined in the voice-mail system's integration file to identify the type of call. The keywords—CGN (calling number), CDN (called number), and FDN (forwarding number)—define the type of call information sent to the voice-mail system.

SUMMARY STEPS

pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

DETAILED STEPS


	Command	Purpose
Step 1	vm-integration Example: `Router(config) vm-integration`	Enters voice-mail integration mode and enables voice-mail integration with DTMF and analog voice-mail system.
Step 2	pattern direct tag1 {CGN \| CDN \| FDN} [tag2 {CGN \| CDN \| FDN}] [tag3 {CGN \| CDN \| FDN}] [last-tag] Example: Router(config-vm-int) pattern direct 2 CGN *	Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone. •The tag attribute is an alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A, B, C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number. The Cisco SRS Telephony router supports a maximum of four tags. •The keywords—CGN, CDN, and FDN—configure the type of call information sent to the voice-mail system, such as calling number (CGN), called number (CDN), or forwarding number (FDN).
Step 3	pattern ext-to-ext busy tag1 {CGN \| CDN \| FDN} [tag2 {CGN \| CDN \| FDN}] [tag3 {CGN \| CDN \| FDN}] [last-tag] Example: Router(config-vm-integration) pattern ext-to-ext busy 7 FDN * CGN *	Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail.
Step 4	pattern ext-to-ext no-answer tag1 {CGN \| CDN \| FDN} [tag2 {CGN \| CDN \| FDN}] [tag3 {CGN \| CDN \| FDN}] [last-tag] Example: Router(config-vm-integration) pattern ext-to-ext no-answer 5 FDN * CGN *	Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
Step 5	pattern trunk-to-ext busy tag1 {CGN \| CDN \| FDN} [tag2 {CGN \| CDN \| FDN}] [tag3 {CGN \| CDN \| FDN}] [last-tag] Example: Router(config-vm-integration) pattern trunk-to-ext busy 6 FDN * CGN *	Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
Step 6	pattern trunk-to-ext no-answer tag1 {CGN \| CDN \| FDN} [tag2 {CGN \| CDN \| FDN}] [tag3 {CGN \| CDN \| FDN}] [last-tag] Example: Router(config-vm-integration) pattern trunk-to-ext no-answer 4 FDN * CGN *	Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.

Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.

Configuring Integration Files on Legacy Voice-Mail Systems

To configure the integration files for a legacy voice-mail system, follow the instructions in the voice-mail system's analog voice mail integration configuration guide or recommended documents. You must design the DTMF integration patterns appropriately, so that the voice-mail system and the Cisco SRS Telephony router work with each other. For a configuration example, see the "Configuring Cisco SRS Telephony for Unity Voice-Mail Integration" section.

Monitoring and Maintaining SRS Telephony

To monitor and maintain the router with SRS Telephony feature, use the following commands in EXEC mode:


Command	Purpose
Router# show run	Displays the configuration.
Router# show call-manager-fallback all	Displays the detailed configuration of all the Cisco IP phones, voice ports, and dial peers of the SRS Telephony router.
Router# show call-manager-fallback dial-peer	Displays the output of the dial peers of the SRS Telephony router.
Router# show call-manager-fallback ephone-dn	Displays Cisco IP phone destination number.
Router# show call-manager-fallback voice-port	Displays output for the voice ports.
Router# show ephone dn dn-tag	Displays Cisco IP phone destination number.
Router# show ephone 7910	Displays Cisco 7910 phone status.
Router# show ephone 7940	Displays Cisco 7940 phone status.
Router# show ephone 7960	Displays Cisco 7960 phone status.
Router# show ephone offhook	Displays Cisco IP phone status for all phones that are off hook.
Router# show ephone registered	Displays Cisco IP phone status for all phones that are currently registered.
Router# show ephone remote	Displays Cisco IP phone status for all nonlocal phones (phones that have no Address Resolution Protocol [ARP] entry).
Router# show ephone ringing	Displays Cisco IP phone status for all phones that are ringing.
Router# show ephone summary	Displays a summary of all Cisco IP phones.
Router# show ephone telephone-number phone-number	Displays Cisco IP phone status for a specific phone number.
Router# show ephone unregistered	Displays Cisco IP phone status for all unregistered phones.
Router# show ephone-dn tag	Displays Cisco IP phone destination number.
Router# show ephone-dn summary	Displays a summary of all Cisco IP phone destination numbers.
Router# show ephone-dn loopback	Displays Cisco IP phone destination numbers in loopback mode.
Router# show voice port summary	Displays a summary of all voice ports.
Router# show dial-peer voice summary	Displays a summary of all voice dial peers.

Configuration Examples

Basic and Optional Configuration Example

This section provides the following configuration example for the basic and optional SRS Telephony configurations described in this chapter:

Voice-Mail Integration Configuration Example

The examples in this section show how to configure analog voice-mail integration. They include configuration examples for local and central voice-mail systems.

Local Voice-Mail System Example

The "Dial-Peer Configuration for Integration for Voice-mail System" section of the example shows a legacy dial-peer configuration for a local voice-mail system. The "Cisco SRS Telephony Voice-mail Integration Pattern Configuration" part is a compatible Cisco SRS Telephony configuration.

Central Location Voice-Mail System Example

The "Dial-Peer Configuration for Integration with Voice-Mail System" section of the example shows a legacy dial-peer configuration for a central voice-mail system. The "Cisco SRS Telephony Voice-mail Integration Pattern Configuration" section is a compatible Cisco SRS Telephony configuration.

Note MWI integration is not supported for PSTN access to voice-mail systems at a central locations.

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