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Table Of Contents
Information About Dialing Plans
Direct Inward Dialing Trunk Lines
Voice Translation Rules and Profiles
How to Configure Dialing Plans
SCCP: Configuring Dial-Plan Patterns
SIP: Configuring Dial-Plan Patterns
Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions
SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions
SCCP: Applying Translation Rules Before Cisco CME 3.2
SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later
SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1
Verifying Voice Translation Rules and Profiles
Activating a Secondary Dial Tone
Configuration Examples for Dialing Plan Features
Feature Information for Dialing Plan Features
Configuring Dialing Plans
Last Updated: March 26, 2007This 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.
Contents
• Information About Dialing Plans
• How to Configure Dialing Plans
• Configuration Examples for Dialing Plan Features
• Feature Information for Dialing Plan Features
Information About Dialing Plans
To design and configure dialing plans, you should understand the following concepts:
• Direct Inward Dialing Trunk Lines
• Voice Translation Rules and Profiles
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" on page 165.
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.
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
Voice translation rules perform manipulations on numbers. Voice translation profiles allow you to group voice translation rules together and associate them with the following:
•Called numbers
•Calling numbers
•Redirected called numbers
Voice translation rules have the ability to perform regular expression matches and replace substrings. The Stream Editor (SED) utility is used to translate numbers. The translation rules replace a substring of the input number if the number matches the match pattern, number plan, and type present in the rule. The SED utility is used to check for a match based on the match pattern.
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.
How to Configure Dialing Plans
This section contains the following tasks:
Dial-Plan Patterns
• SCCP: Configuring Dial-Plan Patterns (required)
• SIP: Configuring Dial-Plan Patterns (required)
• Verifying Dial-Plan Patterns (optional)
Voice Translation Rules
• Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions (required)
• SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions (required)
• SCCP: Applying Translation Rules Before Cisco CME 3.2 (required)
• SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later (required)
• SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1 (required)
• Verifying Voice Translation Rules and Profiles (optional)
Secondary Dial Tone
• Activating a Secondary Dial Tone (optional)
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
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
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 summaryDETAILED 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 summaryUse 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
Cisco CME 3.2 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} voice-translation-rule-tag
8. end
DETAILED STEPS
What to Do Next
•To apply voice translation profiles to SCCP phones connected to Cisco Unified CME 3.2 or a later version, see the "SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions" section.
•To apply voice translation profiles to SIP phones connected to Cisco Unified CME 4.1 or a later version, see the "SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later" section.
•To apply voice translation profiles to SIP phones connected to Cisco Unified CME 3.4 or Cisco Unified 4.0(x), see the "SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1" section.
SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions
To apply a voice translation profile to modify the number dialed by extensions on a SCCP phone, perform the following steps.
Prerequisites
•Cisco CME 3.2 or a later version.
•Voice translation profile containing voice translation rules to be applied must be already configured. For configuration information, see the "Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions" section.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn tag
4. translation-profile {incoming | outgoing} name
5. end
DETAILED STEPS
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" on page 265.
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
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" on page 265.
SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later
To apply a voice translation profile for incoming call legs to a directory number on a SIP phone, perform the following steps.
Prerequisites
•Cisco Unified CME 4.1 or a later version.
•Voice translation profile containing voice translation rules to be applied must be already configured. For configuration information, see the "Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions" section.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. translation-profile incoming name
5. end
DETAILED STEPS
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" on page 270.
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
•Cisco CME 3.4 or a later version.
•Voice translation rule to be applied must be already configured. For configuration information, see the "Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions" section.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag
4. translate-outgoing {called | calling} rule-tag
5. end
DETAILED STEPS
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" on page 270.
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
Configuration Examples for Dialing Plan Features
This section contains the following example:
• Secondary Dial Tone: 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
!
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document TitleCisco Unified CME configuration
• Cisco Unified CME Command Reference
Cisco IOS commands
• Cisco IOS Voice Command Reference
Cisco IOS configuration
• Cisco IOS Voice Configuration Library
Phone documentation for Cisco Unified CME
Technical Assistance
Feature Information for Dialing Plan Features
Table 14 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/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0080189132.html.
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 14 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.
Posted: Wed Aug 15 13:53:18 PDT 2007
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