Configuring the Telephony Session service

This section lists parameters for the Telephony Session service. You set these parameters in Management Station.

The Telephony Session service provides telephony functions like detecting and answering incoming calls, placing calls on hold, transferring, placing outgoing calls, and so on. A key component of the Telephony Session service is the audio provider, which connects the Telephony Session service to audio input and output devices. Voice Platform supports the NMS and SIP audio providers.

This table lists service properties you can set on the Advanced tab for the Telephony Session service.

Service property	Description	App- settable	Data type	Default
audio.nms.CommonSecondaryDevices	Configures the Telephony Session service to use the same ports for incoming calls or outbound bridged or consultation transfers.	No	BOOL	FALSE
audio.nms.DisconnectReason	Returns an error string with the reason why the call disconnected unexpectedly. Not init-settable.	No	STRING	No default
audio.nms.isdncta	Specifies the lines for running the isdncta process to start the D-channel.	Yes	STRING	""
audio.nms.ISDNVariant	Specifies the ISDN protocol variant for the NMS board.	Yes	STRING	""
audio.nms.Lines	Specifies the set of telephony channels to monitor for incoming calls.	No	STRING	Depends on role
audio.nms.OPSCollectDigitnum	Enables the Telephony Session service to collect the ANI/DNIS (caller/called numbers) as DTMF strings.	No	STRING	Disabled
audio.nms.Protocol	Specifies the underlying protocol for the NMS board to use.	Yes	STRING	isd0
audio.nms.RemoteRTPAddressPortStart	Sets the starting IP port number for channels on the telephony board to receive RTP packets.	No	INT	Per Telephony Session service instance (TSS): TSS #1 = 16384 TSS #2 = 16480 TSS #3 = 16576 TSS #4 = 16672
audio.nms.SecondaryDevice	Specifies the secondary line(s) to be used to place the outbound call in the bridged or consultation transfer.	No	STRING	Depends on role
audio.Provider	Specifies the audio provider to use.	No	STRING	sip
audio.sip.AaiBasedOn	Specifies whether information is labelled “Aai:” or “User-to-User:” in the SIP header.	No	STRING	aai
audio.sip.AuthRealmUseridPassword	Specifies a string that the SIP audio provider uses to construct the REGISTER message when a 401 or 407 authentication challenge is received.	No	STRING	""
audio.sip.DisconnectCode	Returns an error string with the code why the call disconnected unexpectedly. Not init-settable.	No	STRING	No default
audio.sip.DNISBasedOn	Specifies the Direct Inward Dialing (DID) routing mechanism used by the SIP audio provider.	No	STRING	ToHeader
audio.sip.FinishNotifyOnBlindTransfer	Tells the gateway to not send Notify requests	No	BOOL	TRUE
audio.sip.Lines	Specifies the range of telephony channels to monitor for incoming calls.	No	STRING	Depends on role
audio.sip.listenipaddress	Specifies the host name or IP address used to build the SIP contact header.	Yes	STRING	IP_address_of local_host
audio.sip.LocationServerURI	Specifies the URI of the SIP location server for the SIP user agent to register itself.	No	STRING	""
audio.sip.ProxyServerURI	Specifies the URI of the SIP proxy server used to send requests.	No	STRING	""
audio.sip.ReferSubOnBlindTransfer	Tells the Telephony Session service to consider the transfer completed as soon as it receives 202 Accepted messages.	No	BOOL	FALSE
audio.sip.SessionExpires	Specifies the time, in seconds, pings are sent and received by both sides in a call. The call is released: If the ping results in an error or no response. If the receiving side fails to get a ping in the specified time.	No	INT	1800
audio.sip.SessionTimerMode	Specifies which side in a call sends pings to determine if other side is active or not. When set to local, the Telephony Session service acts as the pinging side.	No	STRING	local
audio.sip.TransferWaitBeforeBye	In a blind transfer, specifies the time in milliseconds to wait after receiving a 202 Accepted message or the first NOTIFY before sending a BYE.	No	INT	500
audio.sip.UserAgentPort	Port of the SIP user agent listener.	No	STRING	Per Telephony Session service instance (TSS): TSS #1 = 5060 TSS #2 = 5070 TSS #3 = 5080 TSS #4 = 5090
audio.sip.UserAgentURI	The URI for the SIP user agent to register with the location server.	No	STRING	sip:nvp@ %HOSTNAME%:5060
ts.APGetAttributeBufferSize	The buffer size the audio provider uses to obtain application-specific messages in a call.	No	INT	1024
ts.APNumberOfChannels	The number of audio provider channels (or ports) to instantiate.	No	INT	Depends on role
ts.APRedirectsMaxCount	The maximum attempts for the Telephony Session service to redirect a call to a new destination endpoint before abandoning the request.	No	INT	3
ts.APRTPDtmfPayloadType	The payload type ID in the RTP packets sent for DTMF events.	No	INT	101
ts.APRTPLocalIPAddress	The IP address of a second network interface to receive RTP traffic.	No	STRING	IP_address_ local_host
ts.APSIPGatewayList	The SIP telephony gateway to use for transferring or placing outbound calls with tel: URI destinations (not required for sip: URI destinations).	No	STRING	Empty string
ts.APSIPGatewaySupportsForking	Enables Voice Platform to receive audio for hot word recognition from gateway models that support audio forking.	No	BOOL	FALSE
ts.APSIPRedirectsUseGatewayList	Tells the Telephony Session service to look for the gateway of the new destination endpoint address in the list specified by ts.SIPGatewayList.	No	BOOL	TRUE
ts.diagLogPerCall	Creates one diagnostic log file per call (or session).	No	BOOL	FALSE
ts.diagLogPerCompany	Saves diagnostic log files by company name.	No	BOOL	FALSE
ts.DynamicRTPMode	Enables Voice Platform to determine the appropriate forking method based on the gateway model.	No	STRING	DISABLED
ts.EarlyConnectMedia	Enables Voice Platform to stream audio from party C to party A while initiating the transfer.	No	BOOL	FALSE
ts.ForkingGatewayModel	Substring or specific model name and version of a gateway that supports audio forking.	No	STRING	IOS
ts.LogDirectoryHierarchy	Writes diagnostic log files in a hierarchical directory structure that indicates the year, month, day, and hour when the call started.	No	BOOL	FALSE
ts.RTPBridge	Selects RTP or SIP as the communications protocol for incoming and outgoing telephone calls.	No	BOOL	FALSE
ts.ShutdownTimeout	Time, in seconds, the Telephony Session service waits for calls to complete after a shutdown request.	No	INT	-1, meaning disabled
ts.SIPDefaultGateway	The IP address and port number for the Telephony Session service to send SIP messages to the Voice Browser service.	No	STRING	Per Telephony Session service instance (TSS): TSS #1: 127.0.0.1:5062 TSS #2: 127.0.0.1:5072 TSS #3: 127.0.0.1:5082 TSS #4: 127.0.0.1:5092 High-availability roles: TSS #1: ...:5062,6002 TSS #2: ...5072, 6012 TSS #3: ..:5082, 6022 TSS #4: ..:5092, 6032
ts.SIPExtSecurityType	Specifies the allowed versions for SSL and TLS security protocols. Best practice: most deployments change the default to a higher level of security.	No	STRING	TLSv1.0 only
ts.SIPFailedCheckTime	Time, in seconds, between internal OPTIONS polls sent to failed Voice Browser service instances. Used in high-availability roles.	No	INT	30
ts.SIPLocalPort	The port number for Telephony Session service to receive SIP messages from the Voice Browser service.	No	INT	Per Telephony Session service instance (TSS): TSS #1: 5064 TSS #2: 5074 TSS #3: 5084 TSS #4: 5094
ts.SIPOkCheckTime	Time, in seconds, between internal OPTIONS polls sent to functioning Voice Browser service instances. Used in high-availability roles.	No	INT	10
ts.SIPT1	Duration in milliseconds for SIP T1 timer. Used by internal stack to determine INVITE timing. Used in high-availability roles.	No	INT	200
ts.SIPTimerBMultiplier	Multiplier used to calculate the INVITE transaction timeout. Used by internal stack. Used in high-availability roles.	No	INT	30

Configuring the Telephony Session service

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