Policy Name

This is the logical name of the service policy object.

Match Priority

This is only used for policies that match dynamically.

If more than one service policy corresponds with the caller’s user agent name then the one with the highest priority will be used.

SIP End-point Pattern

If this field is not empty the service policy is considered to be dynamically matched and will be attempted for every new call initiated by internal accounts.

This field can contain a full user agent name (e.g. “Linksys/SPA941-5.1.8”) or a comma separated list of patterns (e.g. “Cisco%, Sipura%, Grandstream%”).

NOTE: If the policy is statically assigned to an account, it will always be applied (in spite of the match pattern).

Description

This is a short description of the service policy.

ics

append_incoming_headers

This attribute defines that the call is intended for the mobile network and that PortaSIP® must route the call to the IMS core.

It also defines that PortaSIP® must preserve the ODI (Original Dialog Identifier) of the call when routing it to the IMS. The IMS uses the ODI to match messages to / from PortaSIP® into a single SIP dialog.

Note that the account must not be registered on an IP device. Otherwise, the account registration will override the service policy settings and PortaSwitch® will attempt to terminate the call on a VoIP network.

This attribute is ignored for the calling party.

ro

provision_cld

This attribute allows you to customize the Diameter response sent by PortaBilling® so it includes the Called-Party-Address attribute. The Called-Party-Address is a callee’s ID with translation rules applied.

The default value for this attribute is No since the Called-Party-Address is not required by the majority of gateways.

If your Public data network Gateway (PGW) requires the Called-Party-Address to be included in the response, set the value to Yes.

rise_mscc_result_code

Most gateways connect or terminate calls based on the Result-Code value that is set within the MSCC (Multiple-Services-Credit-Control) AVP. However, some gateways (e. g Huawei ATS 9900) require that the result code be set for the Result-Code AVP.

This attribute allows you to customize the Diameter CCA (Credit-Control-Answer) sent by PortaBilling® so that it copies the Result-Code value from the MSCC AVP and sets it as the Result-Code AVP.

The default value for this attribute is No. To enable this option set it to Yes.

sdp

codec_order_list

Allows the sorting of the codecs list that is configured for a routing plan.

When this attribute is defined, codecs received in the initial INVITE message will be reordered according to the list specified here.

For example, calls to carrier X are sent using the G.711 codec, but you want to send calls using the G.729 codec. You also want to retain the option of sending calls using G.711 in case G.729 is not available.

This attribute is taken from the service policy applied to the calling party; it is ignored if defined for the called party.

Note that this attribute is applicable only when the Enforce codec order option is enabled for the connection on the Routing Filter tab.

sip

always_reliable_1xx

This attribute facilitates early dialog processing with UAs that send malformed provisioning responses to requests with the Supported: 100rel SIP header. That helps prevent stuck sessions with locked funds and allows further service usage.

When this attribute is enabled in dynamically matched service policy, PortaSIP® treats such unreliable responses with SDP as a confirmed answer and proceeds with call processing. Otherwise, PortaSIP® generates the 500 error code and the call is not established.

allow_callee_early_

sdp_change

This attribute allows one-way audio issues caused by SDP changes during call setup to be dealt with successfully.

Possible options are:

• Yes - With this attribute enabled, a B2BUA updates the RTP session if or when an SDP change takes place. This ensures that an appropriate set of parameters is used to establish a media stream.
• No - When this attribute is disabled, B2BUA behaves according to RFC6337 – once the SDP has been received in a SIP response; SDPs in subsequent SIP responses are discarded.

call_progress_filter

Controls the re-sending progress of early media for both participants of a call.

• full_progress - This is the default value. PortaSwitch® just re-sends early media and the 18x call progress responses received from the called party.
• ringing_only - PortaSwitch® turns all 18x call progress responses from the called party into a 180 Ringing message and disables early media for this party only. If defined for the calling party, PortaSwitch® disables any early media in the corresponding call (Music on Waiting and Ring Back Tone will also be disabled).

If this attribute is defined for both calling and called party, the calling party’s options have higher priority and override the options defined for the called party.

call_progress_

notification

Initiates Ring Back Tone generation instead of sending 18x SIP signaling for a call-originating party.

This attribute is ignored for the called party.

Possible options are:

• signaling - This is the default value. PortaSwitch® just re-sends the 18x call progress responses and media received from the called party.
• audio_rbt - PortaSwitch® generates a local ring-back tone when:
• An 18x Ringing response is received without the SDP,
• An 18x Ringing response is received with the SDP, but the RTP media packets are not received within a predefined timeout. Early media (if provided by the called party) is relayed.
• mow - PortaSwitch® plays the Music on Waiting prompt upon receiving a 182 Queued response without the SDP. The rest of the 18x call progress responses are simply resent to the calling party.

callee_stream_

security_settings

Defines the type of media encryption for the called party during the call.

Possible values are:

• as_caller - This is the default setting. PortaSwitch® relays any type of stream received from the calling party. The media features for the account such as Music on Hold, Music on Waiting and Call Recording are not available if the relayed media stream is encrypted.
• decrypted - PortaSwitch® always decrypts / encrypts the media stream for the called party.
• sdes - PortaSwitch® always performs media stream encryption for the called party using the SDES protocol.
• dtls- PortaSwitch® always performs media stream encryption for the called party using the DTLS protocol.
• zrtp - PortaSwitch® always performs media stream encryption for the called party using the zRTP protocol.

caller_stream_

relay_or_decrypt

Defines the types of media encryption and processing for the calling party during the call.

Possible values are:

• forced_relay - PortaSwitch® relays the media stream received from the calling party and ignores the called party’s settings. The media features for the account such as Music on Hold, Music on Waiting and Call Recording are not available if the relayed media stream is encrypted.
• relay_or_decrypt - This is the default setting. PortaSwitch® relays any type of media stream received from the calling party if it is allowed by the called party’s settings. Otherwise, the media stream is encrypted / decrypted.
• decrypt - PortaSwitch® always decrypts the media stream received from the calling party.

early_media_

timeout

fake_180_ringing

first_codec_for_moh

This attribute defines which codes will be used to play the music on hold for the party on hold.

• Yes - PortaSIP® plays music on hold using the codec requested during session renegotiation.
• No - PortaSIP® plays music on hold using an initially negotiated codec.

force_sip_proxy

force_route_through_dsbc

• SIP-UA
• SIP-URI
• outgoing

geo_location_method

gw_longpound_

event

When a user makes a call to an IVR application (e.g. prepaid card, callback pass-through IVR), PortaSIP releases media upon connection. When a user dials a long pound key (##), this event is passed to PortaSIP® in the SIP INFO message.

PortaSIP® uses this input to disconnect the user’s outgoing call leg and continues regular IVR flow processing for the user.

The following options are possible:

• none – PortaSIP ® ignores SIP INFO messages.
• Sonus – PortaSIP ® detects DTMF input in the SIP INFO requests received from the Sonus PSTN gateway and uses it to disconnect the outgoing call leg.

This attribute is used for calls that originated on the Sonus PSTN gateway and works in conjunction with the Release Media Upon Connection option enabled for the IVR application. It is applied to the calling party and is ignored for the party called.

hunt_stop_codes

This attribute is used to adjust call routing for the calling party if:

• A vendor sends an improper SIP response code or
• If a callee does not answer or declines the call.

Recommended values:

• 486 - Busy Here,
• 603 - Declined.

When a connection is tried and PortaSIP® receives a response with a code that matches the one defined, further routing for the call is stopped.

initial_negotiation_

codecs

Defines the codecs list for an SDP generated by PortaSIP® during call pickup if both parties start their calls with a late offer-answer model. If this attribute is defined both for the calling and called parties, common codecs are used. If the list is empty, such pickups are impossible.

Select codecs from the list.

initial_sdp_on_

transfer

This option forces the codec list sent to the transfer target to be the same as the initial codec list from the transferee.

Without this option, the INVITE to the transfer target will contain the last received codec list that was negotiated with the transferee.

keep_alive_interval

no_voice_rejects

This option defines how PortaSIP® responds to a calling party in case of an unauthorized incoming call (e.g. the party called does not exist).

When enabled within the dynamic service policy, PortaSIP® operates as follows, depending upon the option value:

• If set to Yes, PortaSIP® immediately sends a SIP response message (e.g. 404 Not found for non-existent called party), without playing a voice prompt.
• If set to No, PortaSIP® plays the voice prompt and then sends a SIP response message (500 Patience timeout in our example).

old_uri_combining_

for_aor

If set to Yes, PortaSIP® will place the dialed number (CLD) both in the Request-URI (RURI) and To headers in the outgoing INVITE request.

If set to No, the dialed number (CLD) will be placed in the To header only in the outgoing INVITE request.

This attribute can be used in the following scenarios:

• Call to an alias when it is necessary to send an alias number in the RURI instead of the main account number and not just in the “To” header.
• Call to an account where “Simple Forwarding” with “Keep Original CLD” is configured when it is necessary to send an original CLD (forwarder number) in the RURI and not just in the “To” header.

on_hold_media_

direction

This attribute defines the media stream direction when a call is on hold between PortaSIP® and an IP device.

NOTE: The on_hold_media_direction attribute defines the media stream direction only if it is sent through the RTP proxy.

Select one of the available modes:

• sendrecv – This mode enables PortaSIP® to exchange a media stream with an IP device when a call is on hold. This is the recommended option for IP devices located behind the NAT, since it ensures both NAT traversal functionality and the playing of Music on Hold.
• sendonly – This mode enables PortaSIP® only to send a media stream to an IP device when a call is on hold. This is the recommended option for IP devices that are directly reachable from a WAN.
•  auto – This is the default option that automatically defines a media stream direction. If an IP device is located behind the NAT, PortaSIP® sends and receives the media stream (behaves as if in sendrecv mode); if not, PortaSIP® sends only the media stream (behaves as if in sendonly mode).

onjoin_renegotiation_

delay

When a user picks up a call, PortaSIP® starts the session re-negotiation and sends the re-INVITE requests to a vendor. If for some reason the vendor cannot process the re-INVITE request while the previous SIP dialog is still in process (ACK is not yet received), this service policy attribute defines the delay in milliseconds for the re-INVITE sending by PortaSIP®.

This attribute may help fix one-way audio issued during call pickups and other re-INVITE issues. Both calling and called party options are checked; the greater one is used.

passthrough_headers

Use this attribute to configure which headers B2BUA must let through.

For example, B2BUA strips the Alert-Info header that provides an alternative ring tone for the UA from the incoming request because it may introduce the risk of exposing a callee to dubious content if someone were to exploit the URI contained within.

However, if you are sure that the origin of its content is not questionable you may use this header, having defined it for this attribute.

PortaSIP® will accept all the headers listed in this attribute and forward them on to vendors.

remote_address_

sip_field

sdp_cline_mode

Defines how to deal with c-line in outgoing SDP.

The following values are available:

• 0 – c-line will be present only in media sections (no restoring from interval form).
• 1 – c-line will be copied to a common section from the first media section.
• 2 – c-line will be moved to a common section from the first media section and removed from the sections where c-line is the same as in the common one.
• 3 – The same as 0, but also removes c-line from the disabled media sections.
• 4 – The same as 1, but also removes c-line from the disabled media sections.

sdp_ptime_remove

Ptime is the SDP attribute that defines the duration of the media stream in milliseconds for a RTP package.

This attribute instructs PortaSIP® to remove ptime from SDP messages in case a legacy IP device cannot process it.

Possible options are:

• no – Type in this value for ptime to always be available in a SDP message (the default option).
• all – Type in this value to remove ptime from SDP messages.
• <number> – Define the length of media streams in milliseconds; B2BUA removes ptime if it equals this value.

sticky_1xx_sdp

When enabled, B2BUA registers the first SDP received in an 18x SIP response from the callee and sends it to all resulting 18x SIP responses sent to the caller. This is required in those rare cases when a caller UA violates the RFC3261 and after receiving the first 183 SIP response, is unable to process the resulting provision SIP responses (180, 183, etc.) without an SDP. A Service Policy with this attribute must be applied to the caller.

Possible options are:

• Yes
• No

transfer_caller_cli_header

This option specifies the header that is used by PortaSIP to get the CLI for caller authentication. It is only used  for blind transfer scenarios. The header should be specified in the REFER request.

transfer_progress

Triggers Ring Back Tone generation on 18x SIP signaling during Blind Transfer. Transferor’s option is applied.

Possible values are:

• no_indication – PortaSwitch® provides no audio indication during a Blind Transfer.
• transferor_moh – PortaSwitch® plays the Music on Hold prompt when a Blind Transfer is initiated to some destination. Early media from the transfer target is not relayed.
• transferor_moh_or_system – PortaSwitch® plays the Music on Hold prompt (or the default system prompt if the Music on Hold prompt is not selected) when a blind transfer to some destination is initiated. Early media from the transfer target is not relayed.
• ringing_audio – PortaSwitch® generates a local ring-back tone when:
• An 18x Ringing response is received without the SDP.
• oAn 18x Ringing response is received with the SDP, but the RTP media packets are not received within a predefined timeout.

Early media (if sent by the transfer target) is relayed.

sip.identity

out_hdr_contact

Specifies which value must be used for the username part of the URL in the contact SIP header of the outgoing INVITE request.

The following values are available:

• acli – Asserted Calling Line Identification – Provides the actual caller identity if the CLI is hidden or unknown. The caller identity is taken from the PAI (RPID if no PAI) header. For accounts with override identity configured, the identity value can be used.
• cli – Calling Line Identification – The identifier of the calling party, taken from the From header.

out_hdr_diversion

Adds the Diversion SIP header to the outgoing INVITE request and specifies the value to be used for the username part of the header’s URL. This header will be added only if the call is being forwarded and will be ignored if defined for the calling party.

The attribute can have the following value:

• fcli – Calling Line Identification – Redirecting party identity. If it is unavailable, the acli value will be used.

Disable this attribute within the service policy to skip generating Diversion SIP headers.

out_hdr_history

Adds the History-Info SIP header to the outgoing INVITE request and specifies the value to be used for the username part of the header’s URL. This header will be added only if the call is being forwarded and will be ignored if defined for the calling party.

The following values are available:

• fcli – Calling Line Identification – Redirecting party identity.

out_hdr_pai

Adds the P-Asserted-Identity SIP header to the outgoing INVITE request and specifies the value to be used for the username part of the header’s URL.

The following values are available:

• acli – Asserted Calling Line Identification – Provides the actual caller ID information if the CLI is hidden or unknown. The caller ID is taken from the PAI header.
• fcli – Calling Line Identification – Redirecting party identity.

out_hdr_rpid

Adds the Remote-Party-ID SIP header to the outgoing INVITE request and specifies the value to be used for the username part of the header’s URL.

The following values are available:

• acli – Asserted Calling Line Identification – PProvides the actual caller ID information if the CLI is hidden or unknown. The caller ID is taken from the PAI header.
• fcli – Calling Line Identification – Redirecting party identity.

outgoing_headers_trusted

The outgoing_headers_trusted attribute determines whether to include one or both of these headers in a request sent to trusted remote connections.

New attributes specify whether the p-asserted-identity (pai) and remote-party-id (rpid) headers are included in outgoing INVITE requests.

Privacy headers are used when, for example, a caller’s identity must be hidden from a called party, but a vendor still requests this information. In this case, identity information can be included in the privacy header, which indicates that the caller’s info must be withheld from the called party.

Different vendors require and can properly process different privacy headers. Thus, being able to specify which privacy header to send to trusted / untrusted vendors gives you the ability to satisfy both an end user’s request for anonymity and a vendor’s request for identity data.

• pai – Send P-Asserted-Identity.
• rpid – Send Remote-Party-ID.
• none – Do not send any privacy headers.

udptl

t38_fax_max_datagram

t38_fax_udp_ec

Fax error correction scheme. This is used to detect and correct errors in fax page data during reception (e.g. if data is missing or has been altered during transmission).

Possible options are:

• fec – Forward Error Correction – By using the parity-check information that is encoded in the UDPTL packets during fax transmission, PortaSIP® verifies that no data is lost and that none of it contains errors. If any received fax data has become lost or contains errors, PortaSIP® detects it and recovers it when at all possible.

   

  This is the default scheme. The sender’s UA, however, may not support it. In this case, specify redundancy as the fax error correction scheme.
• redundancy – PortaSIP® tries to recover missing fax data using redundant data copies. This mode creates overhead (because of the data copies) for each UDPTL packet, thus more data is sent to PortaSIP®.
• none – PortaSIP® accepts faxes with no error correction.

voice.location

primary_location

• fr.GSTN is the network definition,
• R1R2 is the individual carrier code,
• C1C1C3C4C5 is the city code of the call’s origin, and
• XX are auxiliary digits (00 by default).

audiocodes.cmms

push_events

This attribute enables administrators to control messaging service availability for accounts provisioned in AudioCodes.

When enabled, PortaBilling® sends notification to the AudioCodes CMMS server when a user’s service wallet is empty and their messaging service is suspended. Once a user tops up the service wallet, their messaging service is resumed.

msg

billing_unit_size

bind_as

Defines the mode of connection to the server which determines reception and messages sent:

• Transceiver sends and receives messages from the SMSC over a single SMPP session.
• Transmitter sends short messages to the short message service center (SMSC) and receives the corresponding Short Message Peer-to-Peer (SMPP) responses from the SMSC.
• Receiver receives short messages from the SMSC and returns the corresponding SMPP message responses to the SMSC.

dest_addr_npi

dest_addr_ton

Type of number for destination. The following types are available:

• Unknown
• International
• National
• Network Specific
• Subscriber Number
• Alphanumeric
• Abbreviated

external_routing

If enabled, PortaBilling® will return the routing list for message delivery in the authorization response so that PortaSIP® delivers it to off-net vendors.

If disabled, message will be delivered within the network.

Works in conjunction with the local_lookup option.

heartbeat_interval

incoming_aaa

local_lookup

outgoing_aaa

reconnect_interval

registered_delivery

Numbering Plan Indicator for the SME (Short Message Entity) address where the message originated from.

The possible values are:

• 0 – Do not send delivery reports. This is the default option.
• 1 – Always send delivery reports.
• 2 – Send delivery report in case of an error.
• 3 – Send delivery report only when message is delivered.

reject_error_codes

These are the error codes from your vendor that prevent SMS messages from being resent. Specify the error codes in a hexadecimal format (e.g. 00000058, 58, 0x58).

source_addr_npi

Numbering Plan Indicator for the SME (Short Message Entity) address where the message originated from.

The following indicators are available:

• Unknown
• ISDN
• Data
• Telex
• National
• Privete
• ERMES
• Internet
• WAP Client ID

source_addr_ton

Type of number for source address SME.

The following types are available:

• Unknown
• International
• National
• Network Specific
• Subscriber Number
• Alphanumeric
• Abbreviated

system_type

This attribute categorizes the type of External Short Message Entity (ESME) system that is binding to the SMSC and requests binding it as a receiver.

Specification of the system_type is optional - some SMSCs may not require ESMEs to provide this detail.

transport_protocol

Defines the transport protocol that will be used to deliver the message to the recipient (the default value is set to SIP).

The following protocols are available:

• SIP – Both for on-net and off-net message delivery.
• SMPP – Only for off-net message delivery.

userpart_e164_ translation

If set to Yes, PortaBilling® will use the dialing rules for a phone number translation.

qualution

push_events

This attribute enables administrators to notify end users about their volume discount plan usage via Qualution in real-time.

ro

ro.rise_mscc_result_code

Most gateways deliver messages based on the Result-Code value that is set within the MSCC (Multiple-Services-Credit-Control) AVP. However, some gateways (e. g Huawei ATS 9900) require that the result code be set for the Result-Code AVP.

This attribute allows you to customize the Diameter CCA (Credit-Control-Answer) sent by PortaBilling® so that it copies the Result-Code value from the MSCC AVP and sets it as the Result-Code AVP.

The default value for this attribute is No. To enable this option set it to Yes.