Configure CUBE Recording
Note that all of the following steps should be completed within the context of your overall UCM setup, including basic routing and so on. As part of this configuration you will create a dedicated dial-peer set on the Cisco IOS Router,
To set up CUBE gateway recording for a Verint integration, create a SIP Profile for the PSTN, identifying the dial-peer set within that profile and thereby creating a logical link between the Recorder at the contact center and the calls to be recorded.
Refer to one of the following sections
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Configure recording through CUCM and CUBE Integrated
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Record directly with CUBE
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Record Remote Agents
Configure recording through CUCM and CUBE Integrated
Use the following instructions to configure an environment wherein UCM sends SIP calls to the Recorder, and selects the CUBE/Voice Gateway as the media resource that will forward the RTP traffic.
Configure the following settings via the CUBE Gateway’s IOS (in addition to completing basic configuration according to Cisco’s documentation):
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Allow SIP-to-SIP connections by setting allow-connections to sip to sip.
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Create one dedicated dial-peer to connect to the SIP Trunk on the PSTN PBX from which the calls will come.
Example:
dial-peer voice 1000 voip
description to CUCM10xxA
destination-pattern 10..
session protocol sipv2
session target ipv4:10.199.2.9:5060
voice-class codec 1
dtmf-relay rtp-nte
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Create a second dedicated dial-peer to connect to the SIP Trunk on UCM in the enterprise contact center. The settings should reflect your environment, but it is recommended that you use nte 96 as the rtp payload-type, and sipv2 as the session protocol.
Example:
dial-peer voice 470000 voip
description to CUCM47xxxxATL
destination-pattern 47....
rtp payload-type cisco-codec-fax-ind 126
rtp payload-type nte 96
session protocol sipv2
session target ipv4:10.156.7.220:5060
voice-class codec 1
dtmf-relay rtp-nte
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Configure UCM. Create a new SIP Profile on the UCM PSTN/PBX. When completing the SIP Information screen:
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Assign the address of the dedicated dial-peer set on the Cisco IOS Router as the Destination Address.
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Select the DTMF Signaling Method. Verint recommends RFC 2833.
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Under Device Information, ensure that,
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Media Termination Point Required is not selected
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Retry Video Call as Audio is selected
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Under Recording Information, select This trunk connects to a recording enabled gateway.
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At the DN level for the recording profile above, set the Recording Media Source to Gateway Preferred. With this option selected, behavior will then depend on a few variables. If:
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there is a gateway in the call flow, and the media type (RTP or SRTP) is supported, gateway will be selected.
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the gateway does not support SRTP, phone will be selected.
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there is no gateway in the call flow, phone will be selected.
Please see Cisco’s documentation for additional details and to confirm behavior in a given call flow.
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Create a dedicated SIP Trunk Security Profile with the default settings.
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Provision a SIP Trunk by creating a SIP Trunk on the Device Information page. The SIP Trunk Security Profile (under SIP Information) field must contain the name of the profile created in Step 3 for the PSTN/PBX’s SIP Profile.
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Create another SIP Profile, this one with SIP OPTIONS Ping enabled. This feature is available in CUCM 10 and above. Set:
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Ping Interval for In-service and Partially In-service Trunks (seconds) to 60.
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Ping Interval for Out-of-service Trunks (seconds) to 120
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Ping Retry Timer (milliseconds) to 500
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Ping Retry Count to 6
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Create a new phone Device, with a Device Protocol type SCCP or SIP. Ensure that it uses the following settings:
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Under Device Information:
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Set Built in Bridge to On.
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Set Privacy to On.
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Under Protocol Specific Information, Media Termination Point Required is not selected.
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Record directly with CUBE
Complete the following steps to configure recording directly with CUBE. This configuration does not support recording with CTI correlation. There is no trunk ID or other attribute available that is present in both the SIP trunk call signaling and the JTAPI CTI. For this reason, this solution cannot support CTI.
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Allow SIP-to-SIP connections by setting allow-connections to sip to sip.
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Create one dedicated dial-peer to connect to the SIP Trunk on the PSTN PBX from which the calls will come.
Example:
dial-peer voice 1000 voip
description to CUCM10xxA
destination-pattern 10..
session protocol sipv2
session target ipv4:10.199.2.9:5060
voice-class codec 1
dtmf-relay rtp-nte
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Create a second dedicated dial-peer to connect to the SIP Trunk on UCM in the enterprise contact center. The settings should reflect your environment, but it is recommended that you use nte 96 as the rtp payload-type, and sipv2 as the session protocol.
Example:
dial-peer voice 470000 voip
description to CUCM47xxxxATL
destination-pattern 47....
rtp payload-type cisco-codec-fax-ind 126
rtp payload-type nte 96
session protocol sipv2
session target ipv4:10.156.7.220:5060
voice-class codec 1
dtmf-relay rtp-nte
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Include a media recording profile under the media class for the dial peer (either incoming or outgoing) that is used to route the call through the CUBE. One dial-peer is capable of sending both the audio streams (calling and called) party to the Recorder. In the example below, 6000 and 6001 are the dial peer tags pointing to the Recorder.
Example:
!
media profile recorder 100
media-recording 6000 6001
!
With the media profile set to 100 above, the media class must then be 100 as well.
!media class 100
recorder profile 100
For the dial-peers pointing to the Recorder, use “dummy numbers” as the destination-pattern (rather than numbers from the dial-plan). These dial-peers do not need to include the media-class as they are just used for signaling to and from the media server, and do not need to be recorded.
Example:
dial-peer voice 6000 voip
session target <Recorder IP>
destination-pattern 22222 (dummy number)
session protocol sipv2
dial-peer voice 6001 voip
session target <Recorder IP>
destination-pattern 33333 (dummy number)
session protocol sipv2
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Allow voice connections. Add all PBX and Recorder IPs or IP subnets so that the gateway will trust and allow communication with them over SIP.
Example:
voice service voip
ip address trusted list
ipv4 10.156.7.220
ipv4 10.199.2.9
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Create a codec class enumerating
Example:
voice class codec 1
codec preference 1 g711ulaw
Customers should scope out the dial peer configuration on the CUBE to limit the number of forked SIP calls to Verint. Ideally, only the remote agent RCP call legs are forked. Depending on the configuration, other SIP calls inadvertently forked to Verint may be rejected or recorded without CTI. If a customer cannot scope the CUBE forking, Verint recommends that customers disable the “Record Unknown Devices” setting on the SIP Proxy adapter. With this configuration disabled, Verint will reject any unknown SIP calls, reducing the overall load on the system.
To configure remote agents with CUBE, you must create an IP Extension Pool Member Group in Enterprise Manager, on the data source with which the LCP DNs are associated. The RCP DNs should not be configured in the Verint system; Verint will learn these dynamically through the JTAPI link.
You must also create three adapters: JTAPI, ICM, and SIP Proxy.
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The ICM adapter provides the agent login and state information.
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The JTAPI provides the LCP-RCP pairing and the LCP CTI call information.
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The SIP Proxy serves as the SIP end point for the CUBE to record the calls.
You can also configure the CUBE with a secondary dial peer to leverage 1+1 redundancy and fork the SIP recording to the backup RIS SIP Proxy adapter when the main RIS is down. Note that this redundancy is limited on the CUBE side and appears to only engage when the main RIS SIP Proxy is completely down. The CUBE will not fail over to the backup when the main rejects the SIP call from the CUBE.