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Date Last ModifiedPAGE TITLE
Date Last ModifiedTerminating(active) recording: term refers to the situation when recorded media streams are sent by some device directly to the recorder (e.g. BIB enabled Cisco phones, SIPREC enabled device or similar protocol).
Passive recording: term refers to the situation when recorded media streams are retrieved from the network traffic by monitoring the Mirror Port on the network switch.
Terminating VOIP channels configured in config hw siprec mode.
Following steps assumed to be completed, the prior configuration described in this section:
CLI (Command Line Interface)How to use CLI Tool - refer to CLI Tool Overview |
GUI (Graphical User Interface) |
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Terminating(active) VOIP (hw siprec) channelsFor this example, we assume that Administrator enabled hw siprec in the system. After reload, Administrator can expect following channel list. |
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Open Individual channel in configuration mode. |
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Channel configuration menu can be navigated to through Setup->Channels->SIPREC->Edit
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Display configuration commands available for this channel.
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Channel Configuration Settings |
For details about parameters common to all types of channels - refer to config channel Common Parameters
Command Format | Description | Extra Notes |
deviceid [id] | device id/extension |
device ID or this channel can be mapped to. |
nortptimeout [1-10] | set no rtp timeout (sec). default 3 secs |
Session considered lost if no RTP received during this time |
mapto [y|n] | map to device id/extension | y - map this channel to specific device id (phone) or extension number. Only recordings from this device will be recorded on this channel. "device id" – is a Device Name as specified in the Call Manager e.g. “SEP44D3CA58EC1F”.
n - do not map this channel, the channel is available for dynamic allocation. If caller extension or device id not mapped to a specific channel, it will allocate channel from the pool of unmapped channels and record. If no channels available, call will be rejected. |
gendtmf [0 - 3] | generate DTMF tone according Named Telephone Event (rfc2833) 0- none, 1- near-end, 2- far-end, 3 -all | applicable only for sip protocol |
hndldtmf [0-1] | handle DTMF events 0 - near-end (outbound rtp stream), 1 -all | applicable only for sip protocol |