How to assign a recorded conversation to the final call recipient
IAS 2020.Spring.1 or newer
A new Call Recording setting allows to correlate an incoming, recorded call to the final operator who handled the call.
While using SIPREC recording method, with incoming calls reaching operators through an IVR, it happens that relevant recorded conversation is associated to IVR pilot number or to the first operator involved in the call. Customer wants to see the call associated to the final operator who actually handled the call instead.
A new parameter is available in this setting file:
This parameter controls Cisco SIPREC behaviour: AllowRecordableCallInfoUpdates and it can assume two different values:
- 0 (default): standard behaviour, assigning recording to first DN who answered incoming call.
- 1: Recording is associated to the original User's call recipient (like previous releases), while "User Phone Number" is populated with the final internal phone line who answered the call.
See below sample, where the first call (REC ID 2020...06) has been answered and handled by User "u259" with ext. #259, while second call (REC ID 2020...07) has been forwarded to User "u259" and eventually answered and handled by ext. #9999.
- This setting is relevant only to SIPREC-based recording feature. Other recording methods are not affected.
- In a scenario with multiple call transfers, the recorded conversation will be correlated to the last successful call transfer recipient's DN.
- In case of a conference call, recording is associated to the extension who initiated the conference.
- This feature is taking into account a specific Cisco SIPREC signaling we receive every time the local recipient party changes. This SDP signaling includes the updated list of recipients: IAS is always considering the last recipient, every time there's a new call transfer.
- Above SIPREC signaling does not include custom call headers, like x-incoming, x-outgoing, x-anchor-index, etc.; therefore we are always considering the header included in the original SIP INVITE. This means that Call Direction and participants' order is always evaluated with original header, even after several call transfers.
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