To enable voice dialing, the Cisco CallManager must be configured to route calls to and from the RBX+ app or TEG (Teldio Edge Gateway). The following tasks must be completed, and will be explained in further detail below.
Overview
- Create a SIP Trunk Security Profile for the RBX+ app or TEG.
- Create a SIP Profile
- Configure a SIP Trunk
- Configure a Route Pattern to the RBX+ app or TEG.
The Cisco CallManager configuration must be done through the Cisco System interface. More information about this interface, including how to access it, can be found in the Cisco documentation. The following settings are applicable to most versions of the Cisco CallManager. Consult Cisco Support for further detail on configuration of SIP Trunks.
The RBX system requires a certain amount of SIP Trunks depending on the number of Interconnect Channels licensed. The number of SIP Trunk licenses required will be specified by a Teldio representative.
NOTE: Each Cisco CallManager setup is unique and may require certain parameters to be modified from the values specified below.
The steps to configure the Cisco CallManager are as follows:
Access the Cisco CallManager Administration Webpage
In order to configure a SIP Trunk to communicate with the RBX+ app or TEG, a telecommunications administrator must access the Cisco CallManager administration page.
SIP Trunk Security Profile Configuration
To configure the Security Profile follow the steps below:
- Enter a name for the Security Profile: RBX+ app or TEG.
- Provide a description: Security Profile for RBX+ app or TEG.
- Device Security Mode: Non Secure
- Incoming Transport Type: TCP +UDP
- Outgoing Transport Type: TCP
- Incoming port should be: 5060
Make sure the checkboxes are checked for the options below:
- Accept Presence Subscription
- Accept Out-of-Dialog REFER
- Accept Unsolicited Notification
- Accept Replaces Header
SIP Profile Configuration
Create a SIP Profile which specifies information related to a SIP Peer.
- Enter a name for the SIP Profile: RBX/TEG SIP Profile
- Provide a description for the profile: SIP Profile for the RBX/TEG Server
All default values are acceptable for the SIP Profile.
SIP Trunk Configuration
Create a new SIP Trunk with SIP as the Protocol, then under the Trunk Configuration page set the following parameters:
Device Information
- Enter in a Device Name: RBX/TEG
- Provide a description for the trunk: SIP Trunk to RBX/TEG
- Select a Device Pool for the SIP Trunk to use.
- Common Device Configuration: < None >
- Call Classification: Use System Default
- Media Resource Group List: < None >
- Location: Hub_None
- AAR Group: < None >
- Packet Capture Mode: None
- Packet Capture Duration: 0
Make sure the checkboxes are checked for the options below:
- Media Termination Point Required
- Retry Video Call as Audio
Call Routing Information
- Asserted-Type: Default
- SIP Privacy: Default
Inbound Calls
- Significant Digits: All
- Connected Line ID Presentation: Default
- Connected Name Presentation: Default
Outbound Calls
- Calling Party Selection: Originator
- Calling Line ID Presentation: Default
- Calling Name Presentation: Default
SIP Information
- Destination Address: <IP Address assigned to the RBX/TEG Server>
- Destination Port: 5060
- MTP Preferred Origination Codec: 711ulaw
- Presence Group: Standard Presence group
- SIP Trunk Security Profile: <Select the Security Profile which was created for the RBX/TEG Server>
- SIP Profile: <Select the SIP Profile which was created for the RBX/TEG Server>
- DTMF Signaling Method: No Preference
Route Pattern Configuration
Setup the routing pattern to identify a range of numbers that will be used to forward calls to the RBX+ app or TEG. Ensure the "Gateway/Route List" is set to the RBX/TEG profile created earlier.
- Route Pattern: XXXXX (Example: to forward all 5 digit numbers that begin with 11, enter "11XXX")
- Gateway/Route List: <Select the RBX from the list>
This concludes the configuration necessary for the Cisco CallManager.