Callisto Knowledge Base

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Knowledge Base

SIP Trunk configuration

Table of Contents

Via a Cisco Unified Communications Manager SIP Trunk, incoming calls are forwarded to Callisto, where outgoing calls will be set up.

Add a new SIP Trunk via Device > Trunk

Check Transmit UTF-8 for Calling Party Name

For Encrypted Communication: Check the SRTP option.

At Call Routing Information, check Remote-Party-ID and Asserted-Identity.

At Inbound Calls, check Redirecting Diversion Header Delivery – Inbound.

At Outbound Calls, check Redirecting Diversion Header Delivery – Outbound.

Via SIP Information, enter the Callisto IP address at Destination Address.

For non-secure communication, set Destination Port to 5060.

To encrypt the communication, set this port to 5061.

At SIP Profile, select the profile you have setup in point 1.1.

At SIP Trunk Security Profile, select the profile you have setup in point 1.2.

Info: All Calling Search Space related settings have to be configured according the overall CUCM configuration.