VoIP Trunking On-Net Call

This call type occurs between endpoints connected to Cisco Unified Communications Manager and a Legacy PBX that is connected to a Voice Gateway. The call type includes:

  • SIP/SCCP signaling traffic between the endpoint and the Cisco Unified Communications Manager

  • SIP signaling traffic between the Voice Gateway and the Cisco Unified Communications Manager

  • TDM signaling traffic between the Legacy PBX and Voice Gateway

  • Media traffic between the endpoint and Voice Gateway

On-Net Call (VoIP Trunking)

dialplan-VoIP-Trunking-On-Net

Usage

  • Called number must follow the enterprise internal numbering plan requirement

  • Called number must be defined by ranges and not individual numbers

  • Called number can be either DN or ISP +DN depending on the enterprise internal numbering plan adopted. DN can be either just Extension for flat Dial Plan or SLC + Extension or ISP + SLC + Extension.

  • Voice codec used must be selected per site

  • Only voice calls can be made; video calls are not supported

  • Fax is supported as best effort only

  • MoH is provided in accordance with the site’s MoH policy

  • Voice Gateway configuration is part of the solution

  • Voice Gateway redundant deployment is not supported

  • Enbloc signaling is between the Voice Gateway and Cisco Unified Communication Manager

  • Any target number can be used unless restricted by Class of Service

  • Codec is dynamically selected based on the endpoints used

  • Alternate call routing when the Legacy PBX or Voice Gateway is unreachable is not supported

Accessibility

  • User can perform On-Net call from any endpoint registered with Cisco Unified Communications Manager

  • Legacy PBX connected using a Voice Gateway is considered to be similar to an Inter-Site call

  • User uses the same dialing behavior as Inter-Site On-Net Call

Usage Example

Enables the integration with the existing environment during the transition period of all users

Default Configuration

  • Available to all users at all sites of the enterprise

  • Codec: Voice - G.729 and G.711

  • Codec: Sample Size - 20ms/20Bytes and 20ms/160Bytes

  • Bandwidth: 8kbps and 64kbps

Configuration Choices

  • Feature availability cannot be changed by site or user

  • Codec can be selected between G.711 and G.722 on a per-site basis

Redundancy

Available to users without restrictions

Survivability

Not available to users in fallback mode

Endpoint Types Supported

  • Cisco IP phones

  • Cisco ATA

  • Cisco VG

Example

 
Example:
Phone D (DN=8 200 6100)
Dial 8 400 1234 to Legacy PBX
Connected Number shows 8 400 1234

Legacy PBX (Site 3)
Dial 8 100 2123 to Call A
Phone C (DN=8 100 2123)

previous

Inter-Site On-Net Call

next

Local Gateway (LBO)