5.6 Dial Plan Requirements for MGCP Fallback and Cisco Unified SRST Scenarios

This topic describes the dial plan requirements for MGCP fallback and Cisco Unified SRST.

Cisco Unified SRST mode dial plan should be as close to normal mode as possible:

  • Remote site users need to reach one another by extension.
  • Remote site users need to reach the internal extensions of main site users.
  • Remote site PSTN access should work as usual, including class of service.

SRST failover leaves the remote site independent from the complex dial plan that is implemented in Cisco Unified Communications Manager in the main site. The Cisco Unified SRST router needs to have a minimal dial plan that is implemented to allow for all remote site phones, all main site phones, and all PSTN destinations to be reached with the same numbers as in standard mode.

During fallback, users should be able to dial main site directory numbers as usual. Because these calls have to be routed over the PSTN during fallback, main site extensions have to be translated to E.164 PSTN numbers at the PSTN gateway.

Most enterprises limit the range of destinations that are reachable from specific extensions by applying a class of service to the extensions. This limitation should still be valid during times in SRST mode.

Ensure Connectivity for Remote Sites

This section describes the requirements for the remote site during WAN outage.

PSTN connectivity:

  • You must implement dial peers with destination patterns that correspond to the PSTN access code.
  • Voice translation profiles modify the calling-party number to enable callback.
  • Interdigit timeout adopts open numbering plans that do not have a fixed number of digits.

Intrasite and intersite connectivity:

  • Voice translation profiles expand the called number to PSTN format for site code dialing.
  • The command dialplan-pattern modifies incoming called PSTN numbers to match internally used extensions.

While Cisco Unified SRST is active, you must ensure connectivity from remote sites to PSTN destinations, between different sites, and inside the site itself.

To guarantee PSTN connectivity, you must implement dial peers with destination patterns that correspond to the PSTN access code. In H.323 or SIP gateways, these dial peers must be present for normal operation. When MGCP gateways are used, dial peers are activated by the MGCP gateway fallback mechanism. Interdigit timeout adopts open numbering plans that do not have a fixed number of digits.

Voice translation profiles that are applied to dial peers, the voice interface, or the voice port modify the calling-party ID to enable callback from call lists.

For intrasite and intersite connectivity, voice translation profiles are configured to expand called numbers to the PSTN format during fallback.

The Cisco IOS command dialplan-pattern in the call-manager-fallback configuration mode modifies incoming called numbers to match the remote site extensions. This command also ensures that internal extensions can be dialed even though the lines are configured with the site code and extension. The Line Text Label settings that are defined in Cisco Unified Communications Manager will not be applied to the Cisco Unified SRST phones, so the complete directory number that is applied to the line will be visible to the user.

Ensure Connectivity from Main Site Using CFUR

This section describes the requirements for the main site and other remote sites during a WAN outage.

Remote site has lost connectivity to main site. Phones are registered to the remote gateway.

  • Cisco Unified Communications Manager at the main site does not route calls to the directory numbers of affected IP phones, which are now unregistered in Cisco Unified Communications Manager.
  • CFUR allows routing to alternate numbers for affected (unregistered) IP phones.

During fallback, users of the main site and other remote sites should be able to call remote site users by using their extension numbers.

Cisco Unified Communications Manager considers the remote site phones as unregistered and cannot route calls to the affected IP phone directory numbers. Therefore, if main site users dial internal extensions during the IP WAN outage, the calls will fail (or go to voicemail).

To allow remote IP phones to be reached from other sites, you can configure CFUR at the remote site phones. You should configure the CFUR destination at each remote IP phone with the PSTN number of the IP phone so that internal calls from other sites are forwarded to the PSTN number of an IP phone that is currently unregistered and is therefore not reachable over the IP network.

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