Cisco SRS Telephony Configuration

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

• Prerequisites for SRS Telephony Configuration

• Monitoring and Maintaining SRS Telephony

Prerequisites for SRS Telephony Configuration

The following are prerequisites that must be met before configuration:

•The Cisco SRS Telephony system must be configured with the correct IP phones, platforms, and Cisco CallManager (CCM) version. See the "Cisco IP Phone, Platform, Cisco CallManager, Signal, and Switch Support" section

•The correct memory requirements must be installed. Memory requirements are dependent on the platform and the number of supported Cisco IP phones. See the "Specifications" section for details.

•The correct IP phone firmware must be installed. With CCM V3.1 and later, Cisco SRS Telephony is standard in IP phone firmware. Refer to CCM documentation for firmware requirements. For CCM 3.0.5, you must use the firmware version P003E302 or P004E302.

•IP routing must be enabled.

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

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.

How to Configure Cisco SRS Telephony

This section contains the following procedures:

• Configuring SRS Telephony on Routers to Support IP Phone Functions (required)

• Configuring Cisco SRS Telephony Optional Settings (optional)

• Configuring Cisco SRS Telephony for Unity Voice-Mail Integration (optional)

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

1. call-manager-fallback

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

3. max-dn max-directory-numbers

4. max-ephones max-phones

5. limit-dn {7910 | 7940 | 7960} max-lines

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 "Specifications" 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 "Specifications" 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 1 Enter the show run command to verify the configuration.

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:

a. 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.

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

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

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

Troubleshooting Tips

To troubleshoot the SRS Telephony feature, perform the following steps:

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.

For further debugging, use the debug commands in the Cisco IOS Debug Command Reference.

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:

•Unmatched dial-peer routing

•Cisco IP phone date and time display formats

•Keepalive intervals

•Default destinations for incoming calls

•Global prefixes

•Call transfers from Cisco IP phones to other phone numbers

•Trunk access codes

•Message button phone numbers

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

•Call forwarding during a busy signal or no answer

•Translation rules for numbers dialed on Cisco IP phones

•Interdigit timeout value for all Cisco IP phones attached to the router

•Music on hold

•Dial-peer hunting

•Additional language options for IP phone display

SUMMARY STEPS

1. call-manager-fallback

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

3. date-format {mm-dd-yy | dd-mm-yy}

4. time-format {12 | 24}

5. default-destination telephone number

6. keepalive seconds

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

8. transfer-pattern transfer-pattern

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

10. voicemail phone-number

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

12. call-forward busy directory-number

13. call-forward noan directory-number timeout seconds

14. translate {called | calling} translation-rule-tag

15. timeouts interdigit seconds

16. moh filename

17. user-locale country-code

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 "message URL 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.
Step 17	user-locale country-code Example: Router(config-cm-fallback)# user-locale ES	Selects a language for display on the Cisco IP Phone7940 and Cisco IP Phone 7960. •If you have Cisco CallManager V3.2, enter one of the following ISO-3166 codes: –Denmark—DE –France—FR –Germany—DE –Italy—IT –The Netherlands—NL –Norway—NO –Portugal—PT –Spain—ES –Sweden—SE –United States—US (default) •If you have software prior to Cisco CallManager V3.2, the only country-code option is US (United States).

Troubleshooting Tips

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

•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.

For further debugging, you can use the debug commands in the Cisco IOS Debug Command Reference.

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:

• Configuring DTMF Patterns on the Router (required)

• Configuring Integration Files on Legacy Voice-Mail Systems (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

1. vm-integration

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

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

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

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

6. 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

This section provides the following configuration examples:

• Basic and Optional Configuration Example

• Voice-Mail Integration Configuration Example

Basic and Optional Configuration Example

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

version 12.2

no parser cache

no service single-slot-reload-enable

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

logging rate-limit console 10 except errors

memory-size iomem 30

ip subnet-zero

ip dhcp pool 2600

   network 10.2.0.0 255.255.0.0

   option 150 ip 10.0.0.1

   default-router 10.0.0.1

no ip dhcp-client network-discovery

lcp max-session-starts 0

translation-rule 1

 Rule 0 85... 919785

translation-rule 2

 Rule 0 408734.... 4

interface FastEthernet0/0

 ip address 10.0.0.2 255.255.0.0

 duplex auto

 speed auto

interface FastEthernet0/1

 ip address 10.0.0.1 255.255.0.0

 duplex auto

 speed auto

router eigrp 100

 network 10.0.0.0

auto-summary

 no eigrp log-neighbor-changes

ip classless

ip route 0.0.0.0 0.0.0.0 10.0.0.1

ip http server

snmp-server packetsize 4096

snmp-server manager

call rsvp-sync

voice-port 1/1/0

voice-port 1/1/1

mgcp modem passthrough voip mode ca

no mgcp timer receive-rtcp

mgcp profile default

dial-peer cor custom

 name call911

 name call1800

 name call1900

dial-peer cor list allowall

 member call911

 member call1800

 member call1900

dial-peer cor list allow1800

 member call1800

dial-peer cor list alloww1800and1900

 member call1800

 member call1900

dial-peer cor list allow1900

 member call1900

dial-peer voice 1 voip

 destination-pattern 919715....

 translate-outgoing called 2

 session target ipv4:10.0.0.5

dial-peer voice 2 voip

 shutdown

 destination-pattern 6....T

 session target ipv4:10.0.0.6

 codec g711ulaw

dial-peer voice 3 voip

 destination-pattern 65087.....

 session target ipv4:10.0.0.7

 codec g711ulaw

dial-peer voice 90 voip

 corlist outgoing allow1900

 destination-pattern 9000

 session target ipv4:10.0.0.8

dial-peer voice 45 pots

 destination-pattern 9

 port 1/1/0

call-manager-fallback

 ip source-address 10.0.0.1 port 2000

 max-ephones 10

 max-dn 10

 dialplan-pattern 1 408735.... extension-length 4 no-reg

 dialplan-pattern 2 919785.... extension-length 4 no-reg

 voicemail 4001

 no huntstop

 alias 2 3... to 5555

 translate called 1

 call-forward busy 5001

 call-forward noan 5001 timeout 8

 cor incoming allowall default

 cor incoming allowall 1 4000 - 4999

 cor incoming allowall 2 4000 - 5000

 moh minuet.au

 time-format 12

 date-format mm-dd-yy

 transfer-pattern 1...

 transfer-pattern 2...

 keepalive 30

 interdigit timeout 5

line con 0

line aux 0

line vty 0 4

 login

line vty 5 15

 login

end

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.

! Dial-Peer Configuration for Integration for Voice-mail System

dial-peer voice 101 pots

 destination-pattern 14011

 port 3/0/0

dial-peer voice 102 pots

 preference 1

 destination-pattern 14011

 port 3/0/1

dial-peer voice 103 pots

 preference 2

 destination-pattern 14011

 port 3/1/0

dial-peer voice 104 pots

 destination-pattern A14012

 port 3/1/1

! Cisco SRS Telephony configuration

call-manager-fallback

 max-ephones 24

 max-dn 144

 ip source-address 1.4.214.104 port 2000

 voicemail 14011

! Cisco SRS Telephony Voice-mail Integration Pattern Configuration

vm-integration

 pattern direct 2 CGN *

 pattern ext-to-ext no-answer 5 FDN * CGN *

 pattern ext-to-ext busy 7 FDN * CGN *

 pattern trunk-to-ext no-answer 4 FDN * CGN *

 pattern trunk-to-ext busy 6 FDN * CGN *

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.

! Dial-Peer Configuration for Integration with Voice-Mail System

! located in central location

dial-peer voice 101 pots

 destination-pattern 14011

 port 3/0/0

! Cisco SRS Telephony configuration

call-manager-fallback

 max-ephones 24

 max-dn 144

 ip source-address 1.4.214.104 port 2000

 voicemail 14011

! Cisco SRS Telephony Voice-mail Integration Pattern Configuration

vm-integration

 pattern direct 2 CGN *

 pattern ext-to-ext no-answer 5 FDN * CGN *

 pattern ext-to-ext busy 7 FDN * CGN *

 pattern trunk-to-ext no-answer 4 FDN * CGN *

 pattern trunk-to-ext busy 6 FDN * CGN *

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