Create

Use the following procedure to Create Services Profile:

1. Select the Services tab and click the Create a New Profile () icon in the Main Menu bar. The Create Services Profile dialog is displayed.
   
   
    
2. Enter a Profile Name for this profile, then set the profile values In the Create Services Profile dialog, refer to the Services Attributes table below and fill in the fields as needed to create the Services profile. 
    

   Parameter	Description	Default
   Profile Name	Each profile must have a unique profile name. This value is user-defined; characters can be alphabetic, numeric, a hyphen (-), or an underscore (_). Blank spaces between characters are allowed. Up to 30 characters may be entered. Note: This attribute cannot be edited after the profile has been created.	N/A
   Description	If a profile is given a description, the description must be unique. This value is user-defined; characters can be alphabetic, numeric, a hyphen (-), or an underscore (_). Blank spaces between characters are allowed.	N/A
   MLPP (Multilevel Precedence and Preemption)
   Enable MLPP	The Enable MLPP check-box displays values for MLPP and the Priority Table.
   Priority Table	The priority table defines the r-priority value associated with each MLPP dialed digit. When in MLPP, the dialing prefix is dialed followed by the digit in the priority table. In results in the corresponding r-priority being sent. The defaults are: digit=0, r-priority=8 Flash Override, r-priority value is 8 (90=Flash override, tag msg with 8) digit=1, r-priority=6 Flash, r-priority value is 6 digit=2, r-priority=4 Immediate, r-priority value 4 digit=3, r-priority=2 Priority call, r-priority value is 2 digit=4, r-priority=0, Routine call, r-priority value is 0
   Dialing Prefix	Dialing prefix, one or more digits. The digit table is dialed to activate the MLPP function. By default: 94 - Routine Call 93 - Priority Call 92 - Immediate Call 91 - Flash Call 90 - Flash Override Call	9
   Network Domain	Defines the network domain field of the priority headers on outgoing calls/SIP messages from this line.	uc
   Allowed Domains	Defines the comma separated list of domains allowed. The r-priority from allowed domains is honored.	uc
   Precedence Domain	Defines the precedence domain field of the priority headers. The precedence domain value is 6 characters.	000000
   Max Precedence Level	Defines the max precedence of call allowed. Zero through 9.	0
   QoS DSCP Priority	Defines the DSCP marking for Priority Calls.	47
   QoS DSCP Immediate	Defines the DSCP marking to be used on Immediate Calls.	45
   QoS DSCP Flash	Defines the DSCP marking to be used on Flash Calls.	43
   QoS DSCP Flash Override	Defines the DSCP marking to be used on Flash Override Calls.	41
   QoS DSCP Flash Override Override	Defines the DSCP marking to be used on Flash Override Override Calls. This field is only valid on classified cuc networks.	40
   Media Encapsulation
   RTP/SRTP	Defines how the UA is allowed to communicate media with other UAs. The RTP setting allows RTP encapsulated traffic to be used and is not encrypted. The SRTP setting allows encrypted media traffic between the USa. If both are enabled, the UA selects the most secure method of transfer. If SRTP and RTP are enabled, RTP is used if the far end does not support SRTP.	RTP
   Media
   Codec1	The fields (codec1, codec2, codec3, and codec4) list, in descending order of preference, the audio formats to be negotiated (via SIP) for use by calls on the line. The audio formats are listed in IETF RFC 3555. At present, only codec1 is consulted to determine the order of preference between audio/pcmu and audio/pcma. Syntax: audio.pcmu - PCM mu-law sampled audio audio/pcma - PMC A-law sampled audio. Related Fields: codec2, codec3, codec4	audio/pcmu
   Codec2	This field is currently not used. Syntax: audio.pcmu - PCM mu-law sampled audio audio/pcma - PMC A-law sampled audio. Related Fields: codec1, codec3, codec4	audio/pcma
   Codec3	This field is currently not used. Syntax: audio.pcmu - PCM mu-law sampled audio audio/pcma - PMC A-law sampled audio. Related Fields: codec1, codec2, codec4	N/A
   Codec4	This field is currently not used. Syntax: audio.pcmu - PCM mu-law sampled audio audio/pcma - PMC A-law sampled audio. Related Fields: codec1, codec2, codec3	N/A
   Local Ports Min	The bearer path for Voice over IP (VoIP) calls is sent using Real-time Transport Protocol (RTP) packets which travel over specific User Datagram Protocol (UDP) ports. This field defines a range of UDP ports for use. If not set, the Eqpt uses a default set of ports. In general, this value does not need to be configured unless a specific port range is required. Syntax: Range of UDP ports specified as [lower bound]-[upper-bound], e.g., 53000-54000. Port numbers range from 2 to 65535.
   Local Ports Ma	The bearer path for Voice over IP (VoIP) calls is sent using Real-time Transport Protocol (RTP) packets which travel over specific User Datagram Protocol (UDP) ports. This field defines a range of UDP ports for use. If not set, the Eqpt uses a default set of ports. In general, this value does not need to be configured unless a specific port range is required. Syntax: Range of UDP ports specified as [lower bound]-[upper-bound], e.g., 53000-54000. Port numbers range from 2 to 65535.
   Enable RFC2833 (FAXover VoIP)	When a user attempts a FAX call over a VoIP network, there are two approaches that can be used. ITU T.38 specifies a means of sending the FAX information as demodulated data that is reconstructed at the far end back into the FAX analog waveforms. The other approach is to send the FAX waveforms untouched through the regular RTP bearer path. This field enables the line to attempt to use T.38 transport via SIP protocol negotiation with the far end device. The T.38 approach is considered more robust but may have to be turned off if there are devices in the network that are not properly able to interwork with T.38. Syntax: yes - enables ITU T.38 FAX transport no - disables feature online.	Yes
   Enable RFC2833 (DTMF digits)	Once a call is established and digits are dialed by the user in the talking phase of the call, they are forwarded to the network digit by digit. This may apply when the user has reached their voice mail system or another interactive voice response system which prompts for digits. There are two approaches to sending the DTMF digits into the network. If this field is enabled then it is negotiated with the network to send the digits as IETF RFC 2833 based RTP packets. This is referred to as DTMF relay and is the typical mode of operation in a VoIP network. If this field is not enabled, then the digits are sent within the regular bearer path RTP pay load. In general, this feature should remain enabled on a line unless there is a device in the network that cannot support RFC 2833 and cannot properly indicate that fact through SIP protocol negotiation. Syntax: yes - enable IETF RFC 2833 DTMF relay no - disables this feature online	Yes
   Enable VAD	In VoIP networks there is the capability to not send bearer packets during periods of silence (this termed Silence Suppression). This can reduce the amount of bandwidth consumed within the network. Voice Activity Detection (VAD) plays the role in silence suppression of detecting periods of silence and activity and stopping and starting bearer packet transmission. When bearer packets are not sent, an occasional comfort noise level packet is sent to adjust background noise levels to mask the fact that silence suppression is taking place. This field enables the use of silence suppression (and control noise) on calls (it is re-negotiated with the network via the SIP protocol). Syntax: yes - enables VAD (i.e., silence suppression and comfort noise) no - disables feature online.	No
   QoS DSCP	Both the VoIP bearer (i.e., RTP) packets and control (i.e., SIP) packets have to be assigned proper priority in an IP network. This prevents unnecessary queuing behind lower priority packets. This field represents the DSCP value that populates the DS of RTP (bearer) packets when not in MLPP mode. When MLPP is enabled, this defines the DSCP to be used on Routine Calls. Syntax: This field contains a value from 0 to 63. Related Fields: sip_dscp	35
   SIP
   Registrar URI	A lines Contact Uniform Resource Identifier (URI) is sent by the line to its SIP Registrar (usually the lines softswitch) to allow the Registrar to update its location database as to the current location (Fully Qualified Domain Name (FQDN) or IP address) of the line. This allows incoming calls for the line to be sent to its public address (see address_of_record_uri) and arrive at the lines Proxy Server (i.e., local softswitch serving its domain) which directs the call to the line using the location database. This field represents the URI to send registration requests to (i.e., the Registrar). Syntax: A standard SIP URI (refer to section 19.1 of IETF RFC 3261), i.e., [user]@[host]. In the case of a Registrar URI, the user portion is usually not present and instead only the domain name that the Registrar serves is present, e.g., provider.com:5060. Related Fields: registrar_outbound_proxy_address, contact_uri_user, address_of_record_uri	N/A
   Registrar Out. Pxy Addr	Optional SIP Registrar Outbound Proxy for the line. The Registrar Outbound Proxy is the destination for SIP registration requests sent by the eline. This optional capability, to route SIP registration requests through a specific server, such as a Session Border Controller (SBC), allows a server to manipulate the outgoing requests to support various network functions (e.g., NAT, CALEA, etc.). See outbound_proxy_address for similar behavior for non-registration requests. If a Registrar Outbound Proxy is not defined, then SIP requests usually pass directly to the lines Registrar (i.e., softswitch, see registrar_uri). Syntax: An IP address or fully-qualified domain name (FQDN), e.g., sbc1.provider.com:5060. Related Fields: registrar_uri, ourbound_proxy_address.
   Register Period	A lines Contact Resource Identifier (URI) is sent by the line to its SUP Registrar (usually the lines softswitch) to allow the Registrar to update its location database as to the current location (Fully Qualified Domain Name (FQDN) or IP address) of the line. This allows incoming calls for the line to be sent to its public address (see address_of_record_uri) and arrive at the lines Proxy Server (i.e., local softswitch serving its domain) which directs the call to the line using the local database. This field provides the time (in seconds) that is proposed by the line to the Registrar as the duration of the registration. The Registrar has the final say as to the duration. Syntax: This field is an integer value representing seconds. When set to zero, the line does not re-register. Related Fields: register_head_start, register_retry_interval	3600
   Register Head Start	A lines Contact Uniform Resource Identifier (URI) is sent by the line to its SIP Registrar (usually the lines softswitch) to allow the Registrar to update its location database as to the current location (Fully Qualified Domain Name (FQDN) or IP address) of the line. This allows incoming calls for the line to be sent to its public address (see address_of_record_uri) and arrive at the lines Proxy Server (i.e., local softswitch serving its domain) which directs the call to the line using the location database. This field provides the number of seconds before the previous registration expires at which time the line re-registers. Syntax: This field is an integer value representing seconds. Related Fields: register_period, register_retry_interval.	360
   Register Retry Interval	A lines Contact Uniform Resource Identifier (URI) is sent by the line to its SIP Registrar (usually the lines Softswitch) to allow the Registrar to update its location database as to the current location (Fully Qualified Domain Name (FQDN) or IP address) of the line. This allows incoming calls for the line to be sent to its public address (see address_of_record_uri) and arrive at the lines Proxy Server (i.e., local Softswitch serving its domain) which directs the call to the line using the location database. This field represents the interval (in seconds) between registration retries after a failed registration. Syntax: This field is an integer value representing seconds (0 to 65535). When set to zero, the line does not retry registration. Related Fields: register_period, register_head_start.	60
   Outbound Proxy Address	Optional SIP Registrar Outbound Proxy for the line. The registrar Outbound Proxy is the destination for SIP registration requests sent by the line. This optional capability, to route SIP registration requests through a specific server such as a Session Border Controller (SBC), allows a server to manipulate the outgoing requests to support various network functions (e.g., NAT, CALEA, etc.). See outbound_proxy_address for similar behavior for non-registration requests. If a Registrar Outbound Proxy is not defined, then SIP requests usually pass directly to the lines Registrar (i.e., softswitch), see registrar_uri. Syntax: An IP address or FQDN, e.g., sbc1.provider.com:5060. Related Fields: registrar_uri, outbound_proxy_address.
   Proxy Address	SIP Proxy Server (i.e., softswitch) for the line. The Proxy Server receives incoming calls to the line as they are routed to the lines address of record (AOR, see address_of_record_uri) and the Proxy Server is the destination of the AOR. The Proxy Server then uses the location database, populated via the lines registration contact information, to forward the call to the line. For calls outgoing from the line, the dialed digits act as the user portion of a URI whose host portion is the proxy_address. Outgoing calls therefore pass through the Proxy Server. Note: If a direct_connect_uri is used, then the Proxy Server could be bypassed but usually this uri also uses a host portion that is the same as the proxy_address. Syntax: An IP address or FQDN, e.g., softswitch1.provider.com. Related Fields: outbound_proxy_address, address_of_record_uri, direct_connect_uri.
   Realm	SIP requests from a line may be challenged by the network to provide authentication credentials. A softswitch usually turns authentication on, or off, based on the configuration. The realm identifies which service provider protection domain the line is registered. It is returned in the authentication challenge response. Syntax: This field contains a string of characters. It is often a domain name (e.g., provider.com). Related Fields: username, password.	86400
   Subscribe Period	A SIP SUBSCRIBE message is required to carry out various call features in SIP (e.g., bridged lines). This field provides the time (in seconds) that is proposed by the line to the network as the duration of the subscription. The network has the final say as to the duration. Syntax: This field is an integer value representing seconds. When this field is set to zero, the line does not re-subscribe. Related Fields: subscribe_head_start.
   Subscribe Head Start	SIP SUBSCRIBE message is required to carry out various call features in SIP (e.g., bridged lines). This field provides the number of seconds before the previous subscription expires at which time the line re-subscribes. Syntax: This field is an integer value representing seconds. Related Fields: subscribe_period.
   Obsolete Hold	When a user puts a call on hold, the network is informed of this through SIP. SIP defines two methods to perform this action (see IETF RFC 3264). If this field is enabled then the older scheme to put a call on hold is used. This involves signaling the network that the local IP address for the RTP bearer path is 0.0.0.0. If this field is disabled, then a newer scheme is used which does not change the IP address  to 0.0.0.0. Note: The NORTEL_CS2K softswitch requires that this field be enabled. All Softswitches are required to support the older scheme for backwards compatibility. Syntax: yes - enables older scheme of putting call on hold via 0.0.0.0 IP address no - disables older scheme and uses newer scheme.	No
   Security Model	Defines the security model of the line. If the security model is set to Best Effort, the line is allowed to use the sip: scheme. If the security model is set to Forced Secure, then only the sips: schema is allowed. Currently the AS5300 only supports the sip: schema.	Best Effort
   Session Timers
   Min	All Softswitches employ some periodic auditing mechanism to determine if the calls they believe are active are truly operative on the Eqpt Session timers (refer to IETF RFC 4028) is one such mechanism. Session timers are not supported by all Softswitches, but the feature should be disabled in those cases. This field determines the minimum acceptable value, in seconds, which the line wants a call to be audited. Auditing too often raises the amount of SIP traffic and processing in the network. Syntax: The field contains an integer value representing seconds. Related Fields: session_timer_max, session_timer_refresher.	90
   Max	All Softswitches employ some periodic auditing mechanism to determine if calls they believe are active are truly operative on the Eqpt. Session timers (see IETF RFC 4028) is one such mechanism. Session timers are not supported by all Softswitches, but the feature should work regardless. NORTEL_CS2K and METASWITCH support alternative approaches so this field should be disabled in those cases. This field determines the session timer interval, in seconds, to be proposed by the line to the network. This determines how often a call is audited. Syntax: The field contains an integer value representing seconds. Related Fields: session_timer_min, session_timer_refresher.	1800
   Refresher	All Softswitches employ some periodic auditing mechanism to determine if calls they believe are active are truly operative on the Eqpt. Session timers (see IETF RFC 4028) is one such mechanism. Session timers are not supported by all Softswitches, but the feature should work regardless. NORTEL_CS2K and METASWITCH support alternative approaches so this field and should be disabled in those cases. This field determines if the line requests the use of session timers. If it does then the field also gives the preference as to whether the line or the network initiates the audit. Note: The line always supports session timers if the network chooses to use them whether the line requested their use or not. Syntax: none - the line does not request the use of session timers uas - the line requests their use and prefers it initiates the audit uac - the line requests their use and prefers the network initiates the audit. Related Fields: session_timer_max, session_timer_min.	None
   Dial Plan
   Permanent Timer	When a user goes off-hook and dials digits, the lines digit map (see digit_map) is consulted to determine when dialing is complete. After going off-hook, and before dialing the first digit, the permanent (signal) timer is run. If no digit is entered before the permanent timer expires then digit collection is considered complete and permanent signal treatment is applied to the line (fast busy (i.e., reorder) tone followed by howler (i.e., received off-hook) tone followed by silence. Syntax: 0 (disables permanent timer) or more seconds. Related Fields: digit_map, interdigit_timere, critical_timer, enter_key.	16
   InterDigit Timer	When a user goes off-hook and dials digits, the lines digit map (see digit_map) is consulted to determine when dialing is complete. When at least one digit has been dialed, and a complete pattern match has not yet occurred but some candidate patterns are partially matched and none of them contain the Critical timer, T, at that point, then the (long) interdigit timer is run. If no digit is entered before the (long) interdigit timer expires, then digit collection is considered complete and the collected digits are forwarded to the network. Syntax: 0 (disables interdigit timer) or more seconds. Related Fields: digit_map, permanent_timer, critical_timer, enter_key.	16
   Critical Timer	When a user goes off-hook and dials digits, the lines digit map (see digit_map) is consulted to determine when dialing is complete. When a complete pattern is matched in the digit map and another pattern of longer length may still be matched or if the explicit Critical Timer, T, is arrived at in a matching pattern, then the Critical interdigit timer is started. If no digit is entered before the Critical interdigit timer expires, then digit collection is considered complete and the collected digits are forwarded to the network. Syntax: 0 (immediate timeout) or more seconds. Related Fields: digit_map, permanent_timer, interdigit_timer, enter_key.	4
   Enter Key	When a user goes off-hook and dials digits, the lines digit map (see digit_map) is consulted to determine when dialing is complete. If the user presses the enter key (usually #), and it does not match the current location of any partially matched digit map pattern, then dialing is considered to be complete. Collected digits are then forwarded to the network. Syntax: single character Related Fields: digit_map, interdigit_timer, critical_timer, permanent_timer.	#
   Call Timer
   Timed Release	by the line. This allows the user to go off-hook within the timer period to resume the call. The intended benefit is that if the user fumbles the phone and accidently hangs up then the call is not lost. This field configures the duration of the timer. Syntax: This field contains an integer value, in seconds, for the timed release.

3.  When all the fields have been filled in, click on the Apply button. 
4. The Create Services Profiles dialog remains open. If no additional profiles are to be created, Click the Cancel button. Otherwise, repeat the same procedure to create another Services profile.
5. If no further profiles are to be created, click on the Cancel button to close the Create Services Profile dialog.
6. Click on the Close button again to exit the Profiles display dialog.

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