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       | * Available to all users at all sites of the    |
| Configuration |   enterprise                                    |
|               | * Codec: Voice - G.729 and G.711                |
|               | * Codec: Sample Size - 20ms/20Bytes and         |
|               |   20ms/160Bytes                                 |
|               | * Bandwidth: 8kbps and 64kbps                   |
+---------------+-------------------------------------------------+


+----------------+-------------------------------------------------+
| Configuration  | * Feature availability cannot be changed by     |
| Choices        |   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 | * Cisco IP phones                               |
| Supported      | * 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)                      |
+----------------+-------------------------------------------------+

.. |dialplan-VoIP-Trunking-On-Net| image:: /src/images/dialplan-VoIP-Trunking-On-Net.png