You can point to a route list instead of directly at the SIP trunk, if you wish to configure redundant SIP trunks.Create a route pattern that matches the recording destination address configured in the previous Step.Recording destination address is where the recording calls are sent as shown in the image.Ĭreate Route Pattern to Route Recording Calls.Navigate to Device > Device Settings > Recording Profile.In the example here, the recording server is 14.48.32.170 as shown in the image. The destination address configured is the address of the recording application server.Input the appropriate Device Name, Device Pool, MRGL, SIP trunk security profile, and SIP profile.Create a SIP trunk with the settings as shown in the image.Navigate to Device > Trunk, select Add New.
Create SIP Trunk to Recording Destination
This section describes how to setup the SIP integration of a recording server.
Initiated every time the IP phone user selects the recording option on their IP phone (CUCM 9.x+) or on an application like in this image.The key elements of selective call recording are as follows: The solid line between CUCM and the recording server denotes a CTI connection between CUCM and the application. In the diagram here, the solid lines represent the expected media flow and the dashed lines represent the expected signalling flow. Does not allow recording of phones that are located outside of the managed network (must have access to send RTP directly to recording server).CTI application user must have access to endpoints that need to be recorded.Requires SIP trunk and CTI with recording application.Initiated when the application (recorder) dictates that it must be initiated.Uses BIB of IP phone in order to fork audio to the recording destination.The key elements of application invoked call recording are as follows: In this diagram, the solid lines represent the expected media flow and the dashed lines represent the expected signalling flow:
Does not allow recording of phones that are located outside of the managed network (must have access to send RTP directly to recording server and be a Cisco IP phone capable of allocating a BIB).Some recording vendors require Computer Telephony Integration (CTI) Requires only a SIP trunk between CUCM and recording destination.Initiated every time the IP phone places a call or receives a call.Uses Built-In-Bridge (BIB) of IP phone in order to fork audio to the recording destination.The key elements of automatic call recording are as follows: If your network is live, ensure that you understand the potential impact of any command. All of the devices used in this document started with a cleared (default) configuration. The information in this document was created from the devices in a specific lab environment.
The information in this document is based on these software and hardware versions: Prerequisites RequirementsĬisco recommends that you have knowledge of CUCM integrated with a third-party recording server. This document describes the basics of call recording within Cisco Unified Communications Manager (CUCM), the expected media flow, the expected call flows for Session Initiation Protocol (SIP) and Skinny Client Control Protocol (SCCP) devices, and an example of a common type of call recording setup failure.