CDR Field Mappings

call-account-code

pac

The account code used by a user to make an outbound call. This will default to null unless otherwise configured. More information is available
Account Codes to Restrict Dialing
Unvalidated Account Codes

call-answer-datetime

time-answer

The date-time that this leg of a call was answered (ie. that a SIP 200 OK was sent or received in response to a SIP INVITE).

call-audio-codec

codec

The audio codec that was negotiated for this call, recorded from the call's SDP.

Note: individual phone vendors may use different names for the same codec. You may see "PCMU" or "G.711 u-law" as an example.

call-audio-relay-side-a-local-port

rly_prt_0

Of calls whose RTP was relayed through the Core Server, this is one of the two UDP ports that was used on the Core Server to relay RTP between the orig and term legs of a call.

call-audio-relay-side-a-packet-count

rly_cnt_a

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "A" side of the call.

call-audio-relay-side-a-remote-ip

rly_prt_a

The IP address and UDP port that sent RTP on the "A" side of the call. This may be a remote IP or may be the IP of the Core Server.

This value will be null if the call was never successfully connected.

call-audio-relay-side-b-packet-count

rly_cnt_b

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "B" side of the call.

call-audio-relay-side-b-remote-ip

rly_prt_b

The IP address and UDP port that sent RTP on the "B" side of the call. This may be a remote IP or may be the IP of the Core Server.

This value will be null if the call was never successfully connected.

call-batch-answer-datetime

batch_tim_ans

For all of the batched legs of a single call, this is the date and time that any of the legs was answered (i.e. that a SIP 200 OK was sent or received in response to a SIP INVITE).

call-batch-on-hold-duration-seconds

batch_hold

For all of the batched legs of a single call, this is the accumulated duration in seconds that the call was on hold. This hold duration may be the sum of one or more call legs.

call-batch-sequence-marker

release_code

For all of the batched legs of a single call, this indicates whether this specific leg was to :

• "begin" the batch
• "continue" the batch
• "end" the batch

call-batch-start-datetime

batch_tim_beg

For all of the batched legs of a single call, this is the date and time that the original leg of the call was started on the system.

call-batch-total-duration-seconds

batch_dura

For all of the batched legs of a single call, this is the accumulated duration in seconds of the call, from the the time the call was first answered until the final leg of a call has disconnected.

call-direction

type

Whether the directionality of the call was:

• "Outbound"
• "Inbound"
• "Missed"

call-disconnect-datetime

time_release

The date-time that this leg of a call was disconnected (i.e. that a SIP BYE was sent).

call-disconnect-reason-text

release_text

The text description of the reason this leg of a call was disconnected. Results may include:
"Call completed elsewhere"
"DTMF <3> entered"
"Incompatible Media"
"No Answer"
"Term: Bye" - the term leg of the call sent a SIP BYE

call-disposition

disposition

If Call Dispositions and Reasons have been configured on the system, an agent was logged into one of the graphical interfaces of the system, and the agent completed the Disposition and Reason form after a call, this field will show the Disposition of the call.

More information can be found in this documentation article.

Defaults to null.

call-disposition-notes

notes

If Call Dispositions and Reasons have been configured on the system, an agent was logged into one of the graphical interfaces of the system, and the agent completed the Disposition and Reason form after a call, this field will show any notes the agent may have written for the call.

More information can be found in this documentation article.

Defaults to null.

call-disposition-reason

reason

If Call Dispositions and Reasons have been configured on the system, an agent was logged into one of the graphical interfaces of the system, and the agent completed the Disposition and Reason form after a call, this field will show the Reason for the call's Disposition.

More information can be found in this documentation article.

Defaults to null

call-disposition-submitted-datetime

time_disp

If Call Dispositions and Reasons have been configured on the system, an agent was logged into one of the graphical interfaces of the system, and the agent completed the Disposition and Reason form after a call, this field will show the date-time that the disposition was submitted by the agent.

More information can be found in this documentation article.

Defaults to null

call-disposition-direction

disp_type

If Call Dispositions and Reasons have been configured on the system, an agent was logged into one of the graphical interfaces of the system, and the agent completed the Disposition and Reason form after a call, this field will show whether the call was:

More information can be found in this documentation article.

Defaults to null

call-fax-codec

image_codec

The fax codec that was negotiated for this call, recorded from the call's SDP.

Note: through at least release v44.0, if a fax was transmitted through the system, this field will invariably show "Image" as the codec

call-fax-relay-side-a-local-port

image_prt_0

Of calls whose RTP was relayed through the Core Server, this is one of the two UDP ports that was used on the Core Server to relay RTP between the orig and term legs of a call.

call-fax-relay-side-a-packet-count

image_cnt_a

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "A" side of the call.

call-fax-relay-side-a-remote-ip

image_prt_a

The IP address and UDP port that sent RTP on the "A" side of the call. This may be a remote IP or may be the IP of the Core Server.

This value will be null if the call was never successfully connected.

call-fax-relay-side-b-packet-count

image_cnt_b

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "B" side of the call.

call-fax-relay-side-b-remote-ip

image_prt_b

The IP address and UDP port that sent RTP on the "B" side of the call. This may be a remote IP or may be the IP of the Core Server.

This value will be null if the call was never successfully connected.

call-intelligence-job-id

n/a

This is the id used to tranct the transcription, sentiment and intelegence request to the 3rd party. Needed to request the data.

call-intelligence-percent-negative

n/a

The percent of the call that had a negative sentiment. Only available with intelligence features turned on.

call-intelligence-percent-neutral

n/a

The percent of the call that had a neutral sentiment. Only available with intelligence features turned on.

call-intelligence-percent-positve

n/a

The percent of the call that had a postive sentiment. Only available with intelligence features turned on.

call-intelligence-topics-top

n/a

The top topics discussed during the call. Comma seperated list of topics. Only available with intelligence features turned on.

call-leg-ordinal-index

cdr_index

For all of the batched legs of a single call, this represents what order a specific call leg took place in the batch.

call-on-hold-duration-seconds

time_holding

The duration in seconds that this leg of a call was on hold.

call-orig-call-id

orig_callid

The value of the SIP Call-ID header received in the initial SIP INVITE that started a batch of call legs.

This value will typically be the same Call-ID for all legs in a batch of calls.

call-orig-caller-id

orig_id

The Caller ID for the orig side of a call leg.

call-orig-department

orig_group

The department of the user that initiated the orig leg of a new call. This may be null for calls that originated through a SIP trunk or for users that are not assigned to a department.

call-orig-domain

orig_domain

The domain of the user that initiated the orig leg of a new call. This may be null for calls that originated through a SIP trunk.

call-orig-from-host

orig_from_host

The host portion of the SIP From URI, after passing through initial Dial Translations, just prior to passing through Call Routing.

call-orig-from-name

orig_from_name

The name portion of the SIP From URI, after passing through initial dial translations, just prior to passing through Call Routing.

call-orig-from-uri

orig_from_uri

The complete SIP From URI, after passing through initial Dial Translations, just prior to passing through Call Routing.

call-orig-from-user

orig_from_user

The user portion of the SIP From URI, after passing through initial Dial Translations, just prior to passing through Call Routing.

call-orig-ip-address

orig_ip

The IP address the orig leg of the call was received from.

This may be a remote IP address or may be the IP of the Core Server.

call-orig-match-uri

orig_match

For origination calls (calls inbound to the system), this is the pattern that a SIP header positively matched against to allow the call to begin.

For calls that came in from extensions, this will be the SIP AOR (Address of Record)
(e.g. sip:1000wp@domain)

For calls that came in from SIP trunks, this will be the "Origination Match" pattern of the trunk (e.g. sip*@trunk-name). This includes calls that traversed geo servers in the same cluster.

call-orig-pre-routing-uri

orig_logi_uri

The complete SIP Request URI for the orig leg of a call, prior to passing through initial Dial Translations, and prior to passing through Call Routing.

call-orig-request-host

orig_req_host

The host portion of the SIP URI from the Request line of the SIP INVITE of the orig leg of a call.

call-orig-request-uri

orig_req_uri

The complete SIP URI from the Request line of the SIP INVITE of the orig leg of a call.

call-orig-request-user

orig_req_user

The user portion of the SIP URI from the Request line of the SIP INVITE of the orig leg of a call.

call-orig-reseller

orig_territory

The name of the reseller of the user that initiated the orig leg of a new call. This may be null for calls that originated through a SIP trunk.

call-orig-site

orig_site

The site of the user that initiated the orig leg of a new call. This may be null for calls that originated through a SIP trunk or for users that are not assigned to a site.

call-orig-to-host

orig_to_host

The host portion of the SIP to header of the SIP INVITE of the orig leg of a call.

call-orig-to-uri

orig_to_uri

The complete SIP URI from the SIP To header of the SIP INVITE of the orig leg of a call.

call-orig-to-user

orig_to_user

The user portion of the SIP to header of the SIP INVITE of the orig leg of a call.

call-orig-user

orig_sub

The extension number of the user that initiated the orig leg of a new call. This may be null for calls that originated through a SIP trunk.

call-parent-call-id

servedCallId

The Call-ID that started a new, unique call flow on the system. This field exists but will be empty prior to v44.0

call-parent-cdr-id

cdr_id

The unique ID of this CDR.

call-record-creation-datetime

time_insert

The date-time that this record was inserted into the database.

call-ringing-datetime

time_ringing

The date-time that this leg of a call first started ringing (i.e. that a SIP 180 Ringing or SIP 183 Early Media was sent or received in response to a SIP INVITE).

call-routing-class

route_class

The configured routing class used when the call went through Call Routing. This is likely null in most cases.

call-routing-match-uri

route_to

For all calls on the system that successfully connected one party to another, this is the pattern that a SIP Request URI positively matched against in the system's Call Routing table.

For calls that terminated to extensions, this will be the wildcard Call Route that matches sip:*@*

For calls that terminated through SIP trunks, this will be the "Destination" pattern of the Call Route (e.g. sip:1??????????@*). This includes calls that traversed geo servers in the same cluster.

call-server-mac-address

mac

The MAC address of the Core Server that processed the call.

call-start-datetime

time_start

The date-time that this leg of a call began (i.e. that a SIP INVITE was sent or received).

call-tag

tag

The tag of this leg of a call.

call-talking-duration-seconds

time_talking

The duration in seconds that this leg of a call was in a talking state (does not include the time prior to answer or while on hold).

call-term-call-id

term_callid

The value of the SIP Call-ID header sent in the SIP INVITE of the term leg of the call.

call-term-caller-id

term_id

The Caller ID for the term side of a call leg.

call-term-department

term_group

The department of the user that received. the term leg of the call.

call-term-domain

term_domain

The domain of the user that received the term leg of the call. This may be null for calls that terminated through a SIP trunk .

call-term-ip-address

term_ip

The IP address the term leg of the call was sent to.

This may be a remote IP address or may be the IP of the Core Server.

call-term-match-uri

term_match

For all calls on the system that successfully connected one party to another, this is the pattern that a translated SIP Request URI positively matched against in the system's Connections (SIP Trunk) list or Extensions list.

For calls that terminated to extensions, this will be the AOR of the user sip:1000@domain

For calls that terminated through SIP trunks, this will be the "Termination Match" pattern of the Connection (e.g. sip*@sbc.company.com). This includes calls that traversed geo servers in the same cluster.

call-term-pre-reouting-uri

term_logi_uri

The complete SIP To URI for the term leg of a call, prior to passing through initial Dial Translations, and prior to passing through Call Routing.

call-term-reseller

term_territory

The reseller-territory of the user that received the term leg of the call. This may be null for calls that terminated through a SIP trunk.

call-term-site

term_site

The site of the user that received the term leg of the call. This may be null for calls that terminated through a SIP trunk.

call-term-to-uri

term_to_uri

The complete SIP To URI for the term leg of a call, after passing through initial Dial Translations, and just prior to passing through Call Routing.

call-term-user

term_sub

The extension number of the user that received the term leg of the call. This may be null for calls that terminated through a SIP trunk.

call-through-action

by_action

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-action indicates what system event occurred that furthered the processing of a call. Examples include:

"Call retrieved"
"Forward always"
"Inter-server call leg"
"Queue dispatch"
"Supervised transfer"

call-through-call-id

by_callid

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-call-id is the Call-ID of the call leg that furthered the processing of a call.

call-through-caller-id

by_id

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-caller-id is the Caller ID of the user account of the call leg that furthered the processing of a call.

call-through-department

by_group

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-department is the department of the user account of the call leg that furthered the processing of a call.

call-through-domain

by_domain

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-domain is the domain of the user account of the call leg that furthered the processing of a call

call-through-reseller

by_territory

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-reseller is the reseller-territory of the domain of the user account of the call leg that furthered the processing of a call.

call-through-site

by_site

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-site is the site of the user account of the call leg that furthered the processing of a call.

call-through-uri

by_uri

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-uri is the SIP URI of the call leg that furthered the processing of a call.

call-through-user

by_sub

For call flows more complex than one user calling another, "call-through-" fields identify an intermediary user account that was involved in directing a call from one destination to another.

call-through-user is the user account of the call leg that furthered the processing of a call.

call-total-duration-seconds

duration

The total duration in seconds of this leg of a call. Includes time ringing, talking, and on hold.

call-video-codec

video_codec

The video codec that was negotiated for this call, recorded from the call's SDP.

Note: through at least release v44.0, if video was relayed through the system, this field will invariably show "Video" as the codec.

call-video-relay-side-a-local-port

video_prt_0

Of calls whose RTP was relayed through the Core Server, this is one of the two UDP ports that was used on the Core Server to relay RTP between the orig and term legs of a call.

call-video-relay-side-a-packet-count

video_cnt_a

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "A" side of the call.

call-video-relay-side-a-remote-ip

video_prt_a

The IP address and UDP port that sent RTP on the "A" side of the call. This may be a remote public IP (if video relay was successfully acquired) or a remote private IP (if video relay was not successfully acquired).

This value will be null if video was not negotiated during the call.

call-video-relay-side-b-packet-count

video_cnt_b

Of calls whose RTP was relayed through the Core Server, this is the number of packets that were received from the "B" side of the call.

call-video-relay-side-b-remote-ip

video_prt_b

The IP address and UDP port that sent RTP on the "B" side of the call. This may be a remote public IP (if video relay was successfully acquired) or a remote private IP (if video relay was not successfully acquired).

This value will be null if video was not negotiated during the call.

core-server

hostname

The FQDN of the Core Server that processed the call.

domain

domain

The domain in which this call leg took place.

hide-from-results

hide

Whether this leg of a call should be hidden from the Call History of a user. Typically these are call legs created for internal system processes and not meaningful to an end user.

is-trace-expected

expected_trace

Based on the date the call took place and the Core Server's configured Event Keep Day duration, whether detailed call processing logs should still be available in order to retrieve a call trace (SIP ladder graph) via the API.

prefilled-trace-api

trace_api

A pre-filled v1 API call that can be used to retrieve a call trace. Should only be used if the is-trace-expected parameter is 'yes'.

prefilled-transcription-api

n/a

A pre-filed api link that can be used to get the inteligence and transcription info.

reseller

reseller

The reseller-territory of the domain in which this call leg took place.