Cisco Unified Communications Manager Express System Administrator Guide
Configuring Dialing Plans
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Table of Contents

Configuring Dialing Plans

Contents

Information About Dialing Plans

Phone Number Plan

Dial-Plan Patterns

Direct Inward Dialing Trunk Lines

Voice Translation Rules and Profiles

Secondary Dial Tone

E.164 Enhancements

Phone Registration with Leading + E164 Number

Callback and Calling Number Display

How to Configure Dialing Plans

SCCP: Configuring Dial-Plan Patterns

SIP: Configuring Dial-Plan Patterns

Prerequisites

Verifying Dial-Plan Patterns

Defining Voice Translation Rules in CiscoCME 3.2 and Later Versions

Prerequisites

What to Do Next

SCCP: Applying Voice Translation Rules in CiscoCME 3.2 and Later Versions

Prerequisites

What to Do Next

SCCP: Applying Translation Rules Before CiscoCME 3.2

Prerequisites

What to Do Next

SIP: Applying Voice Translation Rules in CiscoUnifiedCME 4.1 and Later

Prerequisites

What to Do Next

SIP: Applying Voice Translation Rules before CiscoUnifiedCME 4.1

Prerequisites

SUMMARY STEPS

What to Do Next

Verifying Voice Translation Rules and Profiles

Activating a Secondary Dial Tone

Prerequisite

Defining Translation Rules for Callback-Number

Prerequisites

What to Do Next

Examples

Configuration Examples for Dialing Plan Features

Secondary Dial Tone: Example

Voice Translation Rules: Example

Additional References

Related Documents

Technical Assistance

Feature Information for Dialing Plan Features

Configuring Dialing Plans

Last Updated: March 25, 2011

 

This chapter describes features that enable Cisco Unified Communications Manager Express (Cisco Unified CME) to expand or manipulate internal extension numbers so that they conform to numbering plans used by external systems.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a list of the versions in which each feature is supported, see the “Feature Information for Dialing Plan Features” section.

Information About Dialing Plans

To design and configure dialing plans, you should understand the following concepts:

Phone Number Plan

If you install a Cisco Unified CME system to replace an older telephony system that had an established telephone number plan, you can retain the old number plan. Cisco Unified CME supports flexible extension number lengths and can provide automatic conversion between extension dialing and E.164 public telephone number dialing.

When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the full E.164 telephone number) in the call information with the number configured as the destination pattern for the POTS dial peer. The router then strips out the left-justified numbers corresponding to the destination pattern matching the called number. If you have configured a prefix, the prefix will be put in front of the remaining numbers, creating a dial string, which the router will then dial. If all numbers in the destination pattern are stripped-out, the user will receive (depending on the attached equipment) a dial tone.

A successful Cisco Unified CME system requires a telephone numbering plan that supports future expansion. The numbering plan also must not overlap or conflict with other numbers that are on the same VoIP network or are part of a centralized voice mail system.

Cisco Unified CME supports shared lines and multiple lines configured with the same extension number. This means that you can set up several phones to share an extension number to provide coverage for that number. You can also assign several line buttons on a single phone to the same extension number to create a small hunt group. For more information about types of line configurations, see Configuring Phones to Make Basic Calls.

If you are configuring more than one Cisco Unified CME site, you need to decide how calls between the sites will be handled. Calls between Cisco Unified CME phones can be routed either through the PSTN or over VoIP. If you are routing calls over VoIP, you must decide among the following three choices:

  • You can route calls using a global pool of fixed-length extension numbers. For example, all sites have unique extension numbers in the range 5000 to 5999, and routing is managed by a gatekeeper. If you select this method, assign a subrange of extension numbers to each site so that duplicate number assignment does not result. You will have to keep careful records of which Cisco Unified CME system is assigned which number range.
  • You can route calls using a local extension number plus a special prefix for each Cisco Unified CME site. This choice allows you to use the same extension numbers at more than one site.
  • You can use an E.164 PSTN phone number to route calls over VoIP between Cisco Unified CME sites. In this case, intersite callers use the PSTN area code and local prefix to route calls between Cisco Unified CME systems.

If you choose to have a gatekeeper route calls among multiple Cisco Unified CME systems, you may face additional restrictions on the extension number formats that you use. For example, you might be able to register only PSTN-formatted numbers with the gatekeeper. The gatekeeper might not allow the registration of duplicate telephone numbers in different Cisco Unified CME systems, but you might be able to overcome this limitation. Cisco Unified CME allows the selective registration of either 2- to 5-digit extension numbers or 7- to 10-digit PSTN numbers, so registering only PSTN numbers might prevent the gatekeeper from sensing duplicate extensions.

Mapping of public telephone numbers to internal extension numbers is not restricted to simple truncation of the digit string. Digit substitutions can be made by defining dial-plan patterns to be matched. For information about dial plans, see the “Dial-Plan Patterns” section. More sophisticated number manipulations can be managed with voice translation rules and voice translation profiles, which are described in the “Voice Translation Rules and Profiles” section.

In addition, your selection of a numbering scheme for phones that can be directly dialed from the PSTN is limited by your need to use the range of extensions that are assigned to you by the telephone company that provides your connection to the PSTN. For example, if your telephone company assigns you a range from 408 555-0100 to 408 555-0199, you may assign extension numbers only in the range 100 to 199 if those extensions are going to have Direct Inward Dialing (DID) access. For more information about DID, see the “Direct Inward Dialing Trunk Lines” section.

Dial-Plan Patterns

A dial-plan pattern enables abbreviated extensions to be expanded into fully qualified E.164 numbers. Use dial-plan patterns when configuring a network with multiple Cisco Unified CMEs to ensure that the appropriate calling number, extension or E.164 number, is provided to the target Cisco Unified CME, and appears on the phone display of the called phone. In networks that have a single router, you do not need to use dial-plan patterns.

.When you define a directory number for an SCCP phone, the Cisco Unified CME system automatically creates a POTS dial peer with the ephone-dn endpoint as a destination. For SIP phones connected directly into Cisco Unified CME, the dial peer is automatically created when the phone registers. By default, Cisco Unified CME creates a single POTS dial peer for each directory number.

For example, when the ephone-dn with the number 1001 was defined, the following POTS dial peer was automatically created for it:

dial-peer voice 20001 pots

destination-pattern 1001

voice-port 50/0/2

A dial-plan pattern builds additional dial peers for the expanded numbers it creates. If a dialplan pattern is configured and it matches against a directory number, two POTS dial peers are created, one for the abbreviated number and one for the complete E.164 direct-dial telephone number.

For example, if you then define a dial-plan pattern that 1001 will match, such as 40855500.., a second dial peer is created so that calls to both the 0001 and 4085550001 numbers are completed. In this example, the additional dial peer that is automatically created looks like the following:

dial-peer voice 20002 pots

destination-pattern 40855510001

voice-port 50/0/2

In networks with multiple routers, you may need to use dial-plan patterns to expand extensions to E.164 numbers because local extension numbering schemes can overlap each other. Networks with multiple routers have authorities such as gatekeepers that route calls through the network. These authorities require E.164 numbers so that all numbers in the network are unique. Define dial-plan patterns to expand extension numbers into unique E.164 numbers for registering with a gatekeeper. For more information on E.164 numbers, see “E.164 Enhancements” section.

If multiple dial-plan patterns are defined, the system matches extension numbers against the patterns in sequential order, starting with the lowest numbered dial-plan pattern tag first. Once a pattern matches an extension number, the pattern is used to generate an expanded number. If additional patterns subsequently match the extension number, they are not used.

Direct Inward Dialing Trunk Lines

Direct Inward Dialing (DID), is a one-way incoming trunking mechanism, that allows an external caller to directly reach a specific extension without the call being served by an attendant or other intervention.

It is a service offered in which the last few (typically three or four) digits dialed by the caller are forwarded to the called party on a special DID trunk. For example, all the phone numbers from 555-0000 to 555-0999 could be assigned to a company with 20 DID trunks. When a caller dials any number in this range, the call is forwarded on any available trunk. If the caller dialed 555-0234, then the digits 2, 3, and 4 are forwarded. These DID trunks could be terminated on a PBX, so that the extension 234 gets the call without operator assistance. This makes it look as though 555-0234 and the other 999 lines all have direct outside lines, while only requiring 20 trunks to service the 1,000 telephone extensions. Using DID, a company can offer its customers individual phone numbers for each person or workstation within the company without requiring a physical line into the PBX for each possible connection. Compared to regular PBX service, DID saves the cost of a switchboard operator. Calls go through faster, and callers feel they are calling a person rather than a company.

Dial-plan patterns are required to enable calls to DID numbers. When the PSTN connects a DID call for “4085550234” to the Cisco Unified CME system, it also forwards the extension digits “234” to allow the system to route the call.

Voice Translation Rules and Profiles

Translation rules manipulate dialed numbers to conform to internal or external numbering schemes . Voice translation profiles allow you to group translation rules together and apply them to the following types of numbers:

  • Called numbers (DNIS)
  • Calling numbers (ANI)
  • Redirected called numbers
  • Redirected target numbers—These are transfer-to numbers and call-forwarding final destination numbers. Supported by SIP phones in Cisco Unified CME 4.1 and later versions.

After you define a set of translation rules and assign them to a translation profile, you can apply the rules to incoming and outgoing call legs to and from the Cisco Unified CME router based on the directory number. Translation rules can perform regular expression matches and replace substrings. A translation rule replaces a substring of the input number if the number matches the match pattern, number plan, and type present in the rule.

For configuration information, see the “Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions” section.

For examples of voice translation rules and profiles, see the “ Voice Translation Rules ” technical note and the “ Number Translation using Voice Translation Profiles ” technical note.

Secondary Dial Tone

A secondary dial tone is available for Cisco Unified IP phones connected to Cisco Unified CME. The secondary dial tone is generated when a phone user dials a predefined PSTN access prefix and terminates when additional digits are dialed. An example is when a secondary dial tone is heard after a PSTN access prefix, such as the number 9, is dialed to reach an outside line. For configuration information, see the “Activating a Secondary Dial Tone” section.

E.164 Enhancements

Cisco Unified CME 8.5 allows you to present a phone number in + E.164 telephone numbering format. E.164 is an International Telecommunication Union (ITU-T) recommendation that defines the international public telecommunication numbering plan used in the PSTN and other data networks. E.164 defines the format of telephone numbers. A leading + E.164 telephone number can have a maximum of 15 digits and is usually written with a ‘+’ prefix defining the international access code. To dial such numbers from a normal fixed line phone, the appropriate international call prefix must be used.

The leading +E.164 number is unique number specified to a phone or a device. Callers from around the world dial the leading + E.164 phone number to reach a phone or a device without the need to know local or international prefix. The leading + E.164 feature also reduces the overall telephony configuration process by eliminating the need to further translate the telephone numbers.

Phone Registration with Leading + E164 Number

In Cisco Unified CME, phones register using the leading ‘+’ dialing plan in two ways. Phones can either register with the extension number or with leading + E.164 number.

When phones are registered with extension number, the phones will have a dial peer association with the extension number. The dialplan-pattern command is enhanced to allow you to configure leading + phone numbers on the dialplan pattern. Once dialplan-pattern is configured, there could be an E.164 number dialpeer associated with the same phone.

For example, phones registered with extension number 1111 can also be reached by dialing +13332221111. This phone registration method is beneficial in two ways, that is, locally, phones are able to reach each other by just dialing the extension numbers and, remotely, phones can dial abbreviated numbers which are translated as an E.164 number at the outgoing dial-peer. See Example 1 (CME1) for more information.

When phones are registered with a leading + E.164 number, there is only one leading + E.164 number associated with the phone. The demote option in the dialplan-pattern command allows the phone to have two dialpeers associated with the same phone. For more information on configuring the dialplan-patterns, see, How to Configure Dialing Plans .

For example, a phone registered with + E.164 phone number +12223331111 will have two dialpeers associated with the same phone that is, +122233331111 and 1111. See Example 2 (CME2).

Example 1 (CME1)

In the following example, phones are registered with extension number but they can be reached by either dialing the 5-digit extension number, or a leading + E.164 number. When the dial-peer pattern and extension number is configured, phones can also be reached by dialing its + E.164 number. In this example, phone number 41236 (configured in CME 2 Example ) can reach phone number +12223331234 by dialing the abbreviated phone number because the translation profile has the abbreviated rule configured

The phones can reach each other by dialing either the 5-digit extension number or the + E.164 number because the IPv4 address (172.1.1.188) of the phone in CME 2 example is configured in the dial-peer session target for the phone number 41236 in CME 1 example.

 

!

dial-peer voice 333 voip

destination-pattern +1222333....

session target ipv4:172.1.1.188

!

voice translation-rule 1

rule 2 /^3/ /+12223333/

!

voice translation-rule 2

rule 1 /^01555/ /+1555/

!

voice translation-profile abbreviated-rule-1

translate called 1

translate redirect-target 1

!

voice translation-profile callback-rule-2

translate callback-number 2

!

ephone-dn 1

number 41236

translation-profile incoming abbreviated-rule-1

translation-profile outgoing callback-rule-1

!

!

ephone 1

button 1:1

!

!

telephony-service

dialplan-pattern 1 +1333444.... extension-pattern 5

!

voice register dn 1

number 41237

translation-profile incoming abbreviated-rule-1

translation-profile outgoing callback-rule-1

!

!

voice register pool 1

number 1 dn 1

!

voice register global

dialplan-pattern 1 +1333444.... extension-pattern 5

 

Example 2 (CME2)

In the following example, phones are registered with leading + E.164 number and the phones can be reached by dialing either the 5-digit extension number or the + E.164 number. In this example, phone number +12223331234 can reach the phone number 41236 (configured in CME 2 Example). The phone number +12223331234 can reach the phone number 41236 by dialing either the 5-digit extension number or the + E.164 number because the IPv4 address (172.1.1.187) of the phone number 41236 is configured in the dial-peer session target in the CME 2 example.

!

 

dial-peer voice 333 voip

destination-pattern +1333444....

session target ipv4:172.1.1.187

!

voice translation-rule 1

rule 1 /^4/ /+13334444/

!

voice translation-rule 2

rule 1 /^01555/ /+1555/

!

!

voice translation-profile abbreviated-rule-2

translate called 1 translate redirect-target 1

!

!

voice translation-profile callback-rule-2

translate callback-number 2

!

ephone-dn 1

number +12223331234

translation-profile incoming abbreviated-rule-2

translation-profile outgoing callback-rule-2

!

!

ephone 1

button 1:1

!

telephony-service

dialplan-pattern 1 +1222333.... extension-pattern 4 demote

!

voice register dn 1

number +12223331235

translation-profile incoming abbreviated-rule-2

translation-profile outgoing callback-rule-2

!

!

voice register pool 1

number 1 dn 1

!

voice register global

dialplan-pattern 1 +1222333.... extension-pattern 4 demote

Because the legacy phone does not have a ‘+’ button, you can configure dialplan-pattern or translation profile and dial 5 digits.

Let us assume that we have a calling number from PSTN 015556667777 calling to any phone, we know for a fact that the phone number can be translated to a leading + E.164 number as, +15556667777. Then, by applying the translate callback-number above, you can use Local Services or Missed Calls to callback to +15556667777 instead of dialing 015556667777, which is a not universally known number.

Callback and Calling Number Display

In earlier versions of Cisco Unified CME and Cisco Unified SRST, the calling number (number from an incoming call ringing on your phone) was used for both callback (number displayed under Missed Calls in your local phone directory number) and calling numbers. The + E.164 feature in Cisco Unified CME 8.5, allows you to display both calling number and callback numbers in appropriate format so that you are not required to edit the phone numbers before placing a call. The calling number is displayed on the phone when you configure the translation-profile outgoing command in ephone-dn or voice register dn mode.

The translate callback-number configuration in voice translation-profile allows you to translate the callback number and display it in E.164 format. The translate callback number configuration is only applicable for outgoing calls on SIP and SCCP IP phones. When translate callback number is configured, the extra callback field is displayed and if the number matches the translation rule, it is translated. For more information see, “Defining Translation Rules for Callback-Number” section.

Similarly, in Cisco Unified SRST 8.5, you can configure translate calling under voice translation-profile mode to display the calling number. You can configure translation-profile outgoing in call-manager-fallback mode or voice register pool to display the callback number. You can use translate called command in translation-profile and call-manager-fallback or voice register pool will try to match the called number to do the translation. See Enabling Translation Profiles for more information.

The leading ‘+’ in the E.164 number is stripped from the called and calling numbers if the called endpoint or gateway, such as H323 or QSIG gateway, does not support the leading ‘+’ sign in the E.164 number translation. You can strip the leading ‘+’ sign from the number you are calling or a called number using the translation-profile incoming or translation-profile outgoing commands.

How to Configure Dialing Plans

This section contains the following tasks:

Dial-Plan Patterns

Voice Translation Rules

Secondary Dial Tone

E.164 Enhacements

SCCP: Configuring Dial-Plan Patterns

To define a dial-plan pattern, perform the following steps.


Tip In networks that have a single router, you do not need to define dial-plan patterns.


SUMMARY STEPS

1. enable

2. configure terminal

3. telephony-service

4. dialplan-pattern tag pattern extension-length extension-length [ extension-pattern extension-pattern | no-reg ]

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

telephony-service

 

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 4

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

 

Router(config-telephony)# dialplan-pattern 1 4085550100 extension-length 3 extension-pattern 4..

 

Note This example maps all extension numbers 4xx to the PSTN number 40855501xx, so that extension 412 corresponds to 4085550112.

Maps a digit pattern for an abbreviated extension-number prefix to the full E.164 telephone number pattern.

Step 5

end

 

Router(config-telephony)# end

Exits configuration mode and enters privileged EXEC mode.

SIP: Configuring Dial-Plan Patterns

To create and apply a pattern for expanding individual abbreviated SIP extensions into fully qualified E.164 numbers, follow the steps in this section. Dial-plan pattern expansion affects calling numbers and for call forward using B2BUA, redirecting, including originating and last reroute, numbers for SIP extensions in Cisco Unified CME.

Prerequisites

Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register global

4. dialplan-pattern tag pattern extension-length extension-length [ extension-pattern extension-pattern ] [ no-reg ]

5. call-forward system redirecting-expanded

6. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register global

 

Router(config)# voice register global

Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME.

Step 4

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

 

Router(config-register-global)# dialplan-pattern 1 4085550... extension-length 5

Defines pattern that is used to expand abbreviated extension numbers of SIP calling numbers in Cisco Unified CME into fully qualified E.164 numbers.

Step 5

call-forward system redirecting-expanded

 

Router(config-register-global)# call-forward system redirecting-expanded

Applies dial-plan pattern expansion globally to redirecting, including originating and last reroute, numbers for SIP extensions in Cisco Unified CME for call forward using B2BUA.

Step 6

end

 

Router(config-register-global)# end

Exits configuration mode and enters privileged EXEC mode.

Verifying Dial-Plan Patterns

To verify dial-plan pattern configurations, perform the following steps.

SUMMARY STEPS

1. show telephony-service

2. show telephony-service dial-peer
or
show dial-peer summary

DETAILED STEPS


Step 1 show telephony-service

Use this command to verify dial-plan patterns in the configuration.

The following example maps the extension pattern 4.. to the last three digits of the dial-plan pattern 4085550155:

telephony-service

dialplan-pattern 1 4085550155 extension-length 3 extension-pattern 4..

Step 2 SCCP: show telephony-service dial-peer
or
SIP: show dial-peer summary

Use the command to display dial peers that are automatically created by the dialplan-pattern command.

Use this command display the configuration for all VoIP and POTS dial peers configured for a router, including dial peers created by using the dialplan-expansion (voice register) command.

The following example is output from the show dial-peer summary command displaying information for four dial peers, one each for extensions 60001 and 60002 and because the dialplan-expansion command is configured to expand 6.... to 4085555...., one each for 4085550001 and 4085550002. The latter two dial peers will not appear in the running configuration.

Router# show dial-peer summary

AD PRE PASS OUT

TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STATT

20010 pots up up 60002$ 0 0

20011 pots up up 60001$ 0 9

20012 pots up up 5105555001$ 0 9

20013 pots up up 5105555002$ 0 0


 

Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions

To define voice translation rules and voice translation profiles, perform the following steps.


Note To configure translation rules for voice calls in Cisco CME 3.1 and earlier versions, see the "Cisco IOS Voice, Video, and FAX Configuration Guide."


Prerequisites

  • SCCP support—Cisco CME 3.2 or a later version.
  • SIP support—Cisco Unified CME 4.1 or a later version.
  • To define up to 100 translation rules per translation rule table—Cisco Unified CME 8.6 or a later version.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice translation-rule number

4. rule precedence / match-pattern / / replace-pattern /

5. exit

6. voice translation-profile name

7. translate { called | calling | redirect-called | redirect-target } translation-rule-number

8. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice translation-rule number

 

Router(config)# voice translation-rule 1

Defines a translation rule for voice calls and enters voice translation-rule configuration mode.

  • number—Number that identifies the translation rule. Range: 1 to 2147483647.

Step 4

rule precedence / match-pattern / / replace-pattern /

 

Router(cfg-translation-rule)# rule 1 /^9/ //

Defines a translation rule.

  • precedence —Priority of the translation rule. Range: 1 to 100.

Note Range limited to 15 maximum rules in CME 8.5 and earlier versions.

  • match-pattern —Stream Editor (SED) expression used to match incoming call information. The slash (/) is a delimiter in the pattern.
  • replace-pattern —SED expression used to replace the match pattern in the call information. The slash (/) is a delimiter in the pattern.
  •  

Step 5

exit

 

Router(cfg-translation-rule)# exit

Exits voice translation-rule configuration mode.

Step 6

voice translation-profile name

 

Router(config)# voice translation-profile name1

Defines a translation profile for voice calls.

  • name —Name of the translation profile. Maximum length of the voice translation profile name is 31 alphanumeric characters.

Step 7

translate { called | calling | redirect-called | redirect-target } translation-rule-number

 

Router(cfg-translation-profile)# translate called 1

Associates a translation rule with a voice translation profile.

  • called —Associates the translation rule with called numbers.
  • calling —Associates the translation rule with calling numbers.
  • redirect-called —Associates the translation rule with redirected called numbers.
  • redirect-target —Associates the translation rule with transfer-to numbers and call-forwarding final destination numbers. This keyword is supported by SIP phones in Cisco Unified CME 4.1 and later versions.
  • translation-rule-number —Reference number of the translation rule configured in Step 3.
    Range: 1 to 2147483647.

Step 8

end

 

Router(cfg-translation-profile)# end

Returns to privileged EXEC mode.

What to Do Next

SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions

To apply a voice translation profile to incoming or outgoing calls to or from a directory number on a SCCP phone, perform the following steps.

Prerequisites

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone-dn tag

4. translation-profile { incoming | outgoing } name

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn tag

 

Router(config)# ephone-dn 1

Enters ephone-dn configuration mode to create an extension (ephone-dn) for a Cisco Unified IP phone line, an intercom line, a paging line, a voice-mail port, or a message-waiting indicator (MWI).

  • tag —Unique sequence number that identifies this ephone-dn during configuration tasks. Range is 1 to the maximum number of ephone-dns allowed on the router platform. See the CLI help for the maximum value for this argument.

Step 4

translation-profile { incoming | outgoing } name

 
Router(config-ephone-dn)# translation-profile outgoing name1

Assigns a translation profile for incoming or outgoing call legs to or from Cisco Unified IP phones.

  • You can also use an ephone-dn template to apply this command to one or more directory numbers. If you use an ephone-dn template to apply a command and you use the same command in ephone-dn configuration mode for the same directory number, the value that you set in ephone-dn configuration mode has priority.

Step 5

end
 
Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

What to Do Next

If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration file and restart the phones. See Generating Configuration Files for Phones.

SCCP: Applying Translation Rules Before Cisco CME 3.2

To apply a translation rule to an individual directory number in Cisco CME 3.1 and earlier versions, perform the following steps.

Prerequisites

Translation rule to be applied must be already configured by using the translation-rule and rule commands. For configuration information, see the "Cisco IOS Voice, Video, and FAX Configuration Guide."

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone-dn dn-tag

4. translate { called | calling } translation-rule-number

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn tag

 

Router(config)# ephone-dn 1

Enters ephone-dn configuration mode to create directory number for a Cisco Unified IP phone line, an intercom line, a paging line, a voice-mail port, or a message-waiting indicator (MWI).

Step 4

translate { called | calling } translation-rule-tag

 
Router(config-ephone-dn)# translate called 1

Specifies rule to be applied to the directory number being configured.

  • translation-rule-tag —Reference number of previously configured translation rule. Range: 1 to 2147483647.
  • You can use an ephone-dn template to apply this command to one or more directory numbers. If you use an ephone-dn template to apply a command to a directory number and you also use the same command in ephone-dn configuration mode for the same directory number, the value that you set in ephone-dn configuration mode has priority.

Step 5

end
 
Router(cfg-translation-profile)# end

Exits configuration mode and enters privileged EXEC mode.

What to Do Next

If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration file and restart the phones. See Generating Configuration Files for Phones.

SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later

To apply a voice translation profile to incoming calls to a directory number on a SIP phone, perform the following steps.

Prerequisites

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register dn dn - tag

4. translation-profile incoming name

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register dn dn-tag

 

Router(config)# voice register dn 1

Enters voice register dn configuration mode to define a directory number for a SIP phone, intercom line, voice port, or a message-waiting indicator (MWI).

Step 4

translation-profile incoming name

 

Router(config-register-dn)# translation-profile incoming name1

Assigns a translation profile for incoming call legs to this directory number.

Step 5

end

 

Router(config-register-dn)# end

Returns to privileged EXEC mode.

What to Do Next

If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration file and restart the phones. See SIP: Generating Configuration Profiles for SIP Phones.

SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1

To apply an already-configured voice translation rule to modify the number dialed by extensions on a SIP phone, perform the following steps.

Prerequisites

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register pool tag

4. translate-outgoing { called | calling } rule-tag

5. end

DETAILED STEPS

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register pool pool-tag

 

Router(config)# voice register pool 3

Enters voice register pool configuration mode to set phone-specific parameters for SIP phones.

Step 4

translate-outgoing { called | calling } rule-tag

 
Router(config-register-pool)# translate-outgoing called 1

Specifies an already configured voice translation rule to be applied to SIP phone being configured.

Step 5

end

 

Router(config-register-global)# end

Exits configuration mode and enters privileged EXEC mode.

What to Do Next

If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration file and restart the phones. See SIP: Generating Configuration Profiles for SIP Phones.

Verifying Voice Translation Rules and Profiles

To verify voice translation profiles, and rules, perform the following steps.

SUMMARY STEPS

1. show voice translation-profile

2. show voice translation-rule

3. test voice translation-rule

DETAILED STEPS


Step 1 show voice translation-profile [ name ]

This command displays the configuration of one or all translation profiles.

Router# show voice translation-profile profile-8415

Translation Profile: profile-8415

Rule for Calling number: 4
Rule for Called number: 1
Rule for Redirect number: 5
Rule for Redirect-target number: 2

Step 2 show voice translation-rule [ number ]

This command displays the configuration of one or all translation rules.

Router# show voice translation-rule 6

Translation-rule tag: 6

Rule 1:
Match pattern: 65088801..
Replace pattern: 6508880101
Match type: none Replace type: none
Match plan: none Replace plan: none

Step 3 test voice translation-rule number

This command enables you to test your translation rules.

Router(config)# voice translation-rule 5

Router(cfg-translation-rule)# rule 1 /201/ /102/

Router(cfg-translation-rule)# exit

Router(config)# exit

Router# test voice translation-rule 5 2015550101

Matched with rule 5

Original number:2015550101 Translated number:1025550101

Original number type: none Translated number type: none

Original number plan: none Translated number plan: none


 

Activating a Secondary Dial Tone

To activate a secondary dial tone after a phone user dials the specified number string, perform the following steps.

Prerequisite

  • Cisco CME 3.0 or a later version.
  • PSTN access prefix must be configured for outbound dial peer.

SUMMARY STEPS

1. enable

2. configure terminal

3. telephony-service

4. secondary-dialtone digit-string

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

telephony-service

 

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 4

secondary-dialtone digit-string

 

Router(config-telephony)# secondary-dialtone 9

Activates a secondary dial tone when digit-string is dialed.

  • digit-string —String of up to 32 digits that, when dialed, activates a secondary dial tone. Typically, the digit-string is a predefined PSTN access prefix.

Step 5

end

 

Router(config-telephony)# end

Returns to privileged EXEC mode.

Defining Translation Rules for Callback-Number

To define a translation rule for callback numbers on a SIP phone, perform the following steps:

Prerequisites

  • To define up to 100 translation rules per translation rule table—Cisco Unified CME 8.6 or a later version.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice translation-rule number

4. rule precedence | match-pattern | replace-pattern |

5. exit

6. voice translation profile name

7. translate { callback-number | called | calling | redirect-called | redirect-target } translation-rule - number

8. exit

9. voice register pool phone-tag

10. number tag dn dn-tag

11. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice translation-rule number

 

Router(config)# voice translation-rule 10

Defines a translation rule for voice calls and enters voice translation-rule configuration mode.

  • number—Number that identifies the translation rule. Range: 1 to 2147483647.

Step 4

rule precedence | match-pattern | replace-pattern|

 

Router(cfg-translation-rule)# rule 1 /^9/ //

Defines a translation rule.

  • precedence—Priority of the translation rule. Range: 1 to 100.

Step 5

exit

 

Router(cfg-translation-rule)# exit

Exits voice translation-rule configuration mode.

Step 6

voice translation-profile name

 

Router(config)# voice translation-profile eastern

Defines a translation profile for voice calls.

  • name—Name of the translation profile. Maximum length of the voice translation profile name is 31 alphanumeric characters.

Step 7

translate {callback-number | called | calling | redirect-called | redirect-target} translation-rule-number

 

Router(cfg-translation-profile)# translate callback-number 10

Associates a translation rule with a voice translation profile.

  • callback-number—Associates the translation rule with the callback-number.
  • called—Associates the translation rule with called numbers.
  • calling—Associates the translation rule with calling numbers.
  • redirect-called—Associates the translation rule with redirected called numbers.
  • redirect-target—Associates the translation rule with transfer-to numbers and call-forwarding final destination numbers. This keyword is supported by SIP phones in Cisco Unified CME 4.1 and later versions.
  • translation-rule-number—Reference number of the translation rule configured in Step 3. Range: 1 to 2147483647

Step 8

exit

 

Router(cfg-translation-profile))# exit

Exits voice translation-profile configuration mode.

Step 9

voice register pool phone-tag

 

Router(config)# voice register pool 3

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

Step 10

number tag dn dn-tag

 

Router(config-register-pool)# number 1 dn 17

Associates a directory number with the SIP phone being configured.

  • dn dn-tag—Identifies the directory number for this SIP phone as defined by the voice register dn command.

Step 11

end

 

Router(config-translation-profile)# end

Returns to privileged EXEC mode.

What to Do Next

Examples

The following examples shows translation rules defined for callback-number:

!

!

voice service voip

ip address trusted list

ipv4 20.20.20.1

media flow-around

allow-connections sip to sip

!

!

voice translation-rule 10

!

!

voice translation-profile eastcoast

!

voice translation-profile eastern

translate callback-number 10

!

Configuration Examples for Dialing Plan Features

This section contains the following example:

Secondary Dial Tone: Example

telephony-service

fxo hook-flash

load 7910 P00403020214

load 7960-7940 P00305000600

load 7914 S00103020002

load 7905 CP7905040000SCCP040701A

load 7912 CP7912040000SCCP040701A

max-ephones 100

max-dn 500

ip source-address 10.153.233.41 port 2000

max-redirect 20

no service directed-pickup

timeouts ringing 10

system message XYZ Company

voicemail 7189

max-conferences 8 gain -6

moh music-on-hold.au

web admin system name admin1 password admin1

dn-webedit

time-webedit

!

!

!

secondary-dialtone 9

Voice Translation Rules: Example

In the following configuration examples, if a user on Cisco Unified CME 1 dials 94155550100, the call matches on dial peer 9415 and uses translation profile profile-9415 . The called number is translated from 94155550100 to 4155550100, as specified by the translate called command using translation rule 1.

If a user on Cisco Unified CME 1 calls a phone on Cisco Unified CME 2 by dialing 5105550120, and the call forward number is 94155550100, Cisco Unified CME 1 attempts to forward the call to 94155550100. A 302 message is then sent to Cisco Unified CME 1 with the “Contact:” field translated to 4155550100. When the 302 reaches Cisco Unified CME 1, it matches the To: field in the 302 message (5105550120) with dial peer 510. It does incoming translation from 4155550100 to 84155550100, and an INVITE with 84155550100 is sent, which matches dial-peer 8415.

Figure 1 Translation Rules in SIP Call Transfer

 

 

Cisco Unified CME 1 with 408555.... dialplan-pattern
Cisco Unified CME 2 with 510555.... dialplan-pattern

dial-peer voice 9415 voip

translation-profile outgoing profile-9415

destination-pattern 9415555....

session protocol sipv2

session target ipv4:10.4.187.177

codec g711ulaw

voice translation-profile profile-9415

translate called 1

translate redirect-target 1

voice translation-rule 1

rule 1 /^9415/ /415/

dial-peer voice 8415 voip
translation-profile outgoing profile-8415
destination-pattern 8415555....
session protocol sipv2
session target ipv4:10.4.187.177
codec g711ulaw
dial-peer voice 510 voip
translation-profile incoming profile-510
destination-pattern 510555....
session protocol sipv2
session target ipv4:10.4.187.188
codec g711ulaw
voice translation-profile profile-8415
translate called 1
translate redirect-target 2
voice translation-profile profile-510
translate called 3
voice translation-rule 1
rule 1 /^9415/ /415/
voice translation-rule 2
rule 2 /^415/ /9415/
voice translation-rule 3
rule 1 /^8415/ /415/

Additional References

The following sections provide references related to Cisco Unified CME features.

Technical Assistance

Description
Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/techsupport

Feature Information for Dialing Plan Features

Table 1 lists the features in this module and enhancements to the features by version.

To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/33matrix.htm .

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn . An account on Cisco.com is not required.


Note Table 1 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.


 

Table 1 Feature Information for Dialing Plan Features

Feature Name
Cisco Unified CME Versions
Feature Information

Dial-Plan Pattern

4.0

Added support for dial-plan pattern expansion for call forward and call transfer when the forward or transfer-to target is an individual abbreviated SIP extension or an extension that appear on a SIP phone.

2.1

Strips leading digit pattern from extension number when expanding an extension to an E.164 telephone number. The length of the extension pattern must equal the value configured for the extension-length argument.

1.0

Adds a prefix to extensions to transform them into E.164 numbers.

E.164 Enhancements

8.5

Added support for E.164 enhancements.

Secondary Dial Tone

3.0

Support for secondary dial tone after dialing specified number string.

Voice Translation Rules

8.6

Added support for an increased number of translation rules per translaiton table. Old value is 15 maximum, new value is 100 maximum.

4.1

Added support for voice translation profiles for incoming call legs to a directory number on a SIP phone.

3.4

Added support for voice translation rules to modify the number dialed by extensions on a SIP phone.

3.2

Adds, removes, or transforms digits for calls going to or originating from specified ephone-dns.