SIP headers

With this attribute, you can сontrol the direction of the media stream during the early dialog.

By default, the Allow bidirectional early audio attribute is disabled, and PortaSIP only allows unidirectional media transfer from the callee to the caller. Typical examples of early media generated by the callee are ring tone and announcements, for example, queuing status.

Select the checkbox and turn on the Allow bidirectional early audio toggle switch to allow media transfer in both directions.

The Allow bidirectional early audio attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling party and is ignored for the called party.

This attribute allows avoiding one-way audio issues caused by the SDP change during call setup.

By default, Allow callee early SDP change attribute is disabled, so PortaSIP behaves according to RFC6337: once the SDP message has been received in a SIP response, SDP messages in subsequent SIP responses are ignored.

However, some IP PBX equipment (e.g., Samsung OfficeServ 7070 IP PBX) may send a different SDP in subsequent SIP responses. As a result, a one-way audio issue occurs.

When the Allow callee early SDP change attribute is enabled, PortaSIP updates the RTP session when an SDP change takes place. This ensures that an appropriate set of parameters is used to establish a media stream and allows avoiding issues.

To enable this attribute, select the checkbox and turn on the toggle switch.

The Allow callee early SDP change attribute is applied within a service policy assigned to a connection or a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the called party and is ignored for the calling party.

When this attribute is enabled, PortaSIP is allowed to send ICE SDP in a media offer to the callee.

This allows establishing voice calls with User Agents that exclusively use Interactive Connectivity Establishment (ICE) protocol for setting up media streams. Also, it allows RTP streams to traverse network address translation (NAT) devices and firewalls.

The Allow ICE SDP attribute applies within a service policy assigned to a connection or the account (directly or via the product).

By default, when this attribute is disabled, PortaSIP starts call recording only after receiving a positive final response from the callee.

When this attribute is enabled, PortaSIP starts call recording after receiving the early media from the callee.

The Allow recording early media attribute applies within a service policy assigned to a connection, an account (directly or via the product) or a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

The Allow recording early media attribute is used when call recording is enabled. This attribute can be applied to the calling and the called party.

Always reliable 1xx attribute facilitates early dialog processing with User Agents that send malformed provisioning responses to SIP INVITE requests with the Supported: 100rel SIP header. It helps prevent stuck sessions with locked funds and allows further use of the service.

When the Always reliable 1xx attribute is enabled, 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 isn’t established.

To enable this attribute, select the checkbox and turn on the toggle switch.

The Always reliable 1xx attribute applies within a service policy assigned to an account (directly or via the product) or a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be applied to the called party and the calling party.

This attribute controls the re-sending progress of early media for both call participants.

You can select one of the following options from the drop-down list:

• Full progress – this is the default option. PortaSwitch just re-sends early media and the 18x call progress responses received from the called party.

• Ringing only – with this option selected, 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 Ringback Tone will also be disabled).

The Call progress filter attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

If this attribute is defined for both calling and called party, the calling party’s option has higher priority and overrides the option defined for the called party.

This attribute allows to initiate ringback tone generation instead of sending 18x SIP signaling for a calling party.

You can select one of the following options from the drop-down list:

• Signaling – this is the default option. PortaSwitch just re-sends the 18x call progress responses and media received from the called party.

• Audio RBT – PortaSwitch generates a local ringback 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 aren’t received within a predefined timeout. Early media (if provided by the called party) is relayed.

• MOW – PortaSwitch plays the Music on Waiting (MOW) prompt upon receiving a 182 Queued response without the SDP. The rest of the 18x call progress responses are simply re-sent to the calling party.

The Call progress notification attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling party and is ignored for the called party.

This attribute defines the type of media encryption for the called party during the call.

You can select one of the following options from the drop-down list:

• As caller – this is the default option. 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 aren’t 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.

The Callee stream security settings attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the called party and is ignored for the calling party.

This attribute defines how to handle the encrypted stream for the calling party during the call.

You can select one of the following options from the drop-down list:

• 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 aren’t available if the relayed media stream is encrypted.

• Relay or decrypt – this is the default option. 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.

The Caller stream relay or decrypt attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling party and is ignored for the called party.

This attribute defines timeout in seconds during which PortaSwitch waits for RTP media packets upon receiving an 18x Ringing response with the SDP from the called party. If RTP media packets aren’t received within the predefined timeout, PortaSwitch generates its own ringback tone for the caller.

The default timeout is defined by the RTP proxy.

The Early media timeout attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the called party and is ignored for the calling party.

By default, the call redirection fails if an IP address to redirect an incoming call isn’t included in the Endpoint redirect trusted networks list (see the Endpoint redirect trusted networks attribute description below).

Enable this attribute for PortaSIP to send an additional authorization request to PortaBilling to check whether the redirection is allowed for such an IP address. If the authorization is accepted by PortaBilling, PortaSIP redirects the call.

To enable this attribute, select the checkbox for Endpoint redirect action for untrusted networks and select New call authorization.

The Endpoint redirect action for untrusted networks attribute applies within a service policy assigned to an account.

With this attribute, you can specify the list of IP addresses and domain names where PortaSIP is allowed to redirect incoming calls.

For example, PortaSIP receives a 302 (“Moved Temporarily”) response with an IP address of a specific PBX from the redirect server. If the IP address is included in this list, PortaSIP redirects the call to this IP address without additional authorization in PortaBilling.

To add an IP address / domain name to the list:

• select the checkbox for Endpoint redirect trusted networks;

• specify an IP address / domain name;

• to add the next IP address / domain name to the list, click Add .

To delete an IP address / domain name from the list, hover over the IP address / domain name and click Delete .

The Endpoint redirect trusted networks attribute applies within a service policy assigned to an account.

To configure how PortaSIP should act if the received IP address isn’t included in the list of allowed IP addresses / domain names, see the Endpoint redirect action for untrusted networks attribute above.

When this attribute is enabled, PortaSIP sends a “180 Ringing” SIP response to the caller immediately after the originating outgoing INVITE request. By default, when this attribute is disabled, PortaSIP waits for the “180 Ringing” SIP response to be sent by the User Agent of the called party.

To enable this attribute, select the checkbox and turn on the toggle switch.

The Fake 180 ringing attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling party and is ignored for the called party.

This attribute defines which codec is used to play Music on Hold (MOH) for the party on hold.

When the calling or the called party is put on hold during a call, the parties’ User Agents (UAs) re-negotiate the session. Some UAs change their codec list and request the codec that is different from the one initially negotiated for media streaming.

By default, when this attribute is disabled, PortaSIP plays MOH using the codec initially negotiated for media streaming.

When this attribute is enabled, PortaSIP plays MOH using the codec requested during the session re-negotiation. It’s the first codec in the list announced in “200 OK” response by the party that is put on hold. For example, during the call UAs exchange media using codec A. The caller is put on hold and the parties’ UAs re-negotiate the session. The caller’s UA changes the codec order (from "A, B" to "B, A"). Now the caller’s UA requests the first preferred codec – B. PortaSIP uses the codec B and the caller can hear MOH.

To enable this attribute, select the checkbox and turn on the toggle switch.

The First codec for MOH attribute applies within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be applied to the called party and the calling party.

This attribute allows to forbid sending SIP UPDATE requests to the customer’s or vendor’s equipment. For example, when the equipment claims to support SIP UPDATE requests but doesn’t handle it properly.

Enable the Forbid update method attribute to forbid sending SIP UPDATE (e.g., exclude update from the Allow header in the initial INVITE request.)

To enable this attribute, select the checkbox and turn on the toggle switch.

The Forbid update method attribute applies within a service policy assigned to a connection, an account (directly or via the product), or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be applied to the called party and the calling party.

By default, PortaSIP sends originating outgoing INVITE requests to the dispatching SBC (DSBC), and the DSBC then sends the requests to the destinations.

You can change the default settings and send the outgoing calls directly from PortaSIP (bypassing the DSBC). To disable the Force route through DSBC option, select the checkbox and turn off the toggle switch.

You can assign the service policy with this option disabled to:

• outgoing connections “To vendor”;
• the SIP-UA connection to send calls to static IPs of PBXs; and
• the SIP-URI connection to forward calls to external SIP URIs.

This attribute instructs PortaSIP to process the “P-Access-Network-Info” (PANI) SIP header and obtain information about the operator who processes the call and the city where the call originates. According to some European countries’ regulations, e.g., France’s legal requirements, service providers must send this information to their termination partners.

For PortaSIP to check the PANI header in the incoming INVITE request for a wholesale customer, assign the service policy to the corresponding call authorization rule.

For PortaSIP to check the PANI header in the outgoing INVITE request, assign the service policy to the vendor connection.

To enable this attribute, select the check box and add a geo location method on the panel that opens.

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 an IVR application.

When a user makes a call to an IVR application (e.g., prepaid card calling application), 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 can use this input to disconnect the user’s outgoing call leg and continue regular IVR flow processing for the user.

You can select one of the following options from the drop-down list:

• None – PortaSIP ignores SIP INFO messages. This is the default option.

• 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.

The GW long pound event attribute applies within a service policy assigned to a connection. It’s applied to the calling party that makes an outgoing call to an IVR.

The attribute is ignored for the called party.

With this attribute, you can define the hunt stop codes list. The hunt stop codes allow to adjust call routing for the calling party if:

• a vendor sends an improper SIP response code; or

• if a callee doesn’t answer or declines the call.

When a connection is tried and PortaSIP receives a response with a code that matches the one specified in the hunt stop codes list, further routing for the call is stopped.

To add hunt stop codes to the list:

• select the checkbox for the Hunt stop codes attribute;

• specify a code;

• to add the next code to the list, click Add .

To delete a code from the list, hover over the code and click Delete .

The recommended codes are:

• 486 - Busy Here,

• 603 - Declined.

The Hunt stop codes attribute applies within a service policy assigned to a “To vendor” connection.

Some mobile operators may not accept requests from nodes that don’t include the “user=phone” parameter to the outgoing requests. When this attribute is enabled, it adds the “user=phone” parameter to the Remote-Party-Id/P-Asserted-Identity headers URI.

To enable this attribute, select the checkbox and turn on the toggle switch.

The Include 'user=phone' in PAI, RPID URI attribute applies within a service policy assigned to a connection or account (directly or via the product).

This attribute is applied to the called party and is ignored for the calling party.

This attribute allows you to define the codec list for an SDP message that PortaSIP generates during call pickup if both parties start their calls with a late offer-answer model.

The following scenarios are possible:

1. If this attribute is defined both for the calling and called party, the common codecs are used.

2. If this attribute is defined both for the calling and called party and there are no common codecs, PortaSIP will try to negotiate a session using codecs defined for the calling party.

3. If this attribute is defined for only one party (either calling or called), PortaSIP will try to negotiate a session using codecs that are defined in this attribute.

4. If this attribute isn’t defined for both parties such pickups will be rejected.

To select the codecs, select the Initial negotiation codecs checkbox and go to the corresponding panel that automatically opens.

The Initial negotiation codecs attribute is applied within a service policy assigned to an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

When a call is transferred, the INVITE request is sent to the transfer destination. By default, when this attribute is disabled, this INVITE request contains the last received codec list that was negotiated with the transferred party.

When this attribute is enabled, the INVITE request to the transfer destination contains the SDP with initial codec list from the transferred party.

The Initial SDP on transfer attribute is applied within a service policy assigned to an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be applied to the calling party and the called party.

Using this attribute, you can specify the interval in seconds for keepalive call monitoring. When this attribute is applied this interval will be used to send SIP requests to the calling/called party. By default, this interval is 90 seconds and it’s the minimum value that can be applied.

To disable keep-alive requests, specify “0” value.

The Keepalive interval attribute applies within a service policy assigned to a connection, an account (directly or via the product), or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be individually defined for the calling and the called party.

The keepalive functionality supports three SIP termination response codes by default: 408 (Request Timeout), 481 (Call Leg/Transaction Does Not Exist), or 486 (Busy Here). When PortaSIP receives these codes, it terminates the call.

To add more termination codes to the list:

• select the checkbox for the Keepalive termination codes attribute;

• specify a code, e.g., 503 (Service Unavailable);

• to add the next code, e.g., 603 (Declined) to the list, click Add .

To delete a SIP termination response code from the list, hover over the code and click Delete .

Note that a response with one of these codes to the first keep-alive SIP request will not terminate the call, as it means that the calling/called party doesn't support keepalives.

The Keepalive termination codes attribute applies within a service policy assigned to a connection or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute can be applied to the calling and the called party.

The No response timeout attribute indicates the time in seconds after which the attempt to reach a destination is discontinued unless a real response (any response with a code higher than 100) is received from a vendor. If the attribute value is set to 0, the call time to the destination is limited only by the expire timer.

To specify the attribute value, select the checkbox for the No response timeout attribute.

The No response timeout attribute applies within a service policy assigned to an outgoing connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute defines how PortaSIP responds to a calling party in case of an unauthorized incoming call (e.g., the party called doesn’t exist).

By default, when this attribute is disabled, PortaSIP plays the voice prompt and then sends a SIP response message (e.g., 500 Patience timeout).

When this attribute is enabled, PortaSIP immediately sends a SIP response message (e.g., 404 Not found for non-existent called party), without playing a voice prompt.

To enable the No voice rejects attribute, select the checkbox and turn on the toggle switch.

This attribute may be useful for those vendors who want to reduce processing time or re-route the call in case of failure.

The No voice rejects attribute applies within a service policy assigned to an incoming connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute defines the media stream direction between PortaSIP and an IP device when a call is on hold. Select the needed attribute mode for your IP device to ensure that Music on Hold is functioning correctly.

You can select one of the following modes:

• Auto – this is the default mode that automatically defines the media stream direction based on the NAT settings: if an IP device is located behind the NAT, PortaSIP sends and receives the media stream (behaves as if in Send and receive mode); if not, PortaSIP only sends the media stream (behaves as if in Send only mode).

• Send and receive – enable this mode for PortaSIP to exchange the 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.

• Send only – enable this mode for PortaSIP to only 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.

The On hold media direction attribute applies within a service policy assigned to a connection.

This attribute can be applied to the calling and the called party.

When a user picks up a call, PortaSIP starts the session re-negotiation and sends re-INVITE requests to a vendor. This attribute defines the delay in milliseconds for PortaSIP to send the re-INVITE request. The delay allows to avoid situations when a User Agent receives re-INVITE earlier than 200 OK response for the previous INVITE. Thus, this attribute may help fix one-way audio issues during call pickups and other re-INVITE issues.

If this attribute is disabled for both calling and called party, the default value 100 milliseconds is applied. If this attribute is enabled for the calling/called party or both, the greater value is applied.

The Onjoin renegotiation delay attribute applies within a service policy assigned to a connection or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

In outgoing INVITE requests, PortaSIP fills the "To" header with the dialed number (target CLD). This attribute allows you to configure what value PortaSIP uses to fill the Request-URI (RURI) in outgoing INVITE requests. Depending on the selected mode, PortaSIP can use the target CLD or the routing number (translated according to dialing rules) that are sent in an authorization response from PortaBilling.

With this attribute, you can also enable PortaSIP to pass the number portability parameters during an outgoing call.

Select one of the following options from the drop-down list. Note that the default value is Internal RURI.

• RURI – PortaSIP fills the "To" header with the CLD and the RURI with the routing number.

  This option can be used if a remote vendor expects that this vendor’s ID is sent in RURI while the originally dialed number is sent in the "To" header.

• AOR – PortaSIP fills both the "To" header and the RURI with the CLD. Also, PortaSIP adds "PortaSIP-Notify" header with the "aor=" parameter filled with the routing number. This option can be used in the following scenarios:

  • Calls 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.

  • Calls 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.

• NPDI – PortaSIP fills both the "To" header and the RURI with the CLD. Also, PortaSIP adds two parameters to the RURI:

  • rn – the routing number value defines a target prefix which is used instead of the originally dialed number to authorize, rate, and route the call; it contains the value defined in the origin field in the Number_Portability table.

  • npdi – PortaSIP adds the "yes" value that indicates that number lookup in the PortaSwitch local ported numbers database was made.

  When you use this option, a vendor can determine that a number lookup in the PortaSwitch local ported numbers database is made and whether a number is ported.

• Internal RURI – this is the default value. PortaSIP checks whether the route contains "gwname" parameter with the value "SIP-UA", "SIP-URI" or "INTERNAL". If yes, PortaSIP behaves as if the RURI option is selected for this attribute. Otherwise, PortaSIP fills both the "To" header and the RURI with the CLD.

The Out routing number mode attribute applies within a service policy assigned to a connection or an account (directly or via the product). This attribute is applied to the called party and is ignored for the calling party.

PortaSIP usually strips unknown or potentially unsafe headers from incoming requests. This attribute allows you to configure which headers PortaSIP must let through.

For example, PortaSIP strips the “Alert-Info” header that provides an alternative ring tone for a User Agent 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 isn’t questionable you can make PortaSIP use this header by specifying it within this attribute.

To configure the header list:

• select the checkbox to enable the Passthrough headers attribute;

• specify a header name;

• to add the next header to the list, click Add ;

• to delete a header from the list, hover over the header name and click Delete .

PortaSIP will accept all the headers listed within this attribute and forward them to vendors.

The Passthrough headers attribute applies within a service policy assigned to a connection, call authorization rule or an account (directly or via the product).

This attribute is applied to the calling party.

When an incoming call arrives from an external network and the “P-Access-Network-Info” (PANI) header in the INVITE request is absent or invalid, PortaSIP can override the PANI with a valid one. This attribute allows you to specify the caller location information that will be used in the PANI header for incoming calls to comply with EU regulations.

To specify the Primary location attribute value, use the GSTNR1R2C1C1C3C4C5XX pattern, for example, GSTN771234500.

• GSTN is the default network definition;

• R1R2 is the service provider’s individual code;

• C1C1C3C4C5 is the city code of the call origin; and

• XX are auxiliary digits (00 by default).

The Primary location attribute applies within a service policy assigned to an incoming connection.

This option allows you to relay the location, e.g., received in the multipart SDP message from an IP phone/SIP client, to the vendor that handles emergency calls. See the example of the call scenario in the Route emergency calls and send user location data from MS Teams chapter.

The Relay location info attribute applies within a service policy assigned to a connection or the account (directly or via the product).

This attribute is applied to the called party and is ignored for the calling party.

When this attribute is enabled, PortaSIP relays 18x responses with new SDP messages to the caller during subsequent routing hunting. Thus, you can configure PortaSIP to inform the caller’s User Agent (UA) about the new RTP port without the session update. This attribute may be useful for UAs that don’t support the UPDATE requests.

To enable the Relay subsequent 18x attribute, select the checkbox and turn on the toggle switch.

The Relay subsequent 18x attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

With this attribute, you can specify the name of a SIP header from which PortaSIP should get the remote IP to pass this IP to PortaBilling in h323-remote-address Radius attribute.

The Remote address SIP field attribute applies within a service policy assigned to call authorization rule.

Enable this attribute for PortaSIP to remove video section attributes from an SDP message when a video codec is disabled in the Routing filters (VoIP from/to vendor connection > Service configuration > Routing filters) or prohibited in the Codec policy (Routing plan > Codec policy).

By default, when this attribute is disabled, PortaSIP only sends “0” port in m=video header to disable the media stream and doesn’t remove video section attributes from the SDP message.

This attribute is useful when you use IP phones (e.g., Polycom VVX500) that don’t indicate that the media stream is disabled when the port is set to “0” and the video section attributes are present in the SDP. With this attribute disabled, such IP phones may attempt to display video when it's not needed, e.g., when calling auto attendant.

The Remove SDP video section attribute applies within a service policy assigned to a connection or an account (directly or via a product).

This attribute can be applied to the called party and the calling party.

By default, when this attribute is disabled, the SIP headers that are generated in the PortaBilling authorization response are passed through PortaSIP.

You can change the default behavior so that the headers will not be passed through PortaSIP even if listed in the Passthrough headers service policy attribute. These headers are: History-Info, Diversion, Privacy, P-Asserted-Identity, Remote-Party-Id, P-Preferred-Identity, Cisco-GUID, h323-conf-id, h323-incoming-conf-id. For this, select the checkbox for the Restrict PortaBilling controlled headers attribute, and turn on the toggle.

The Restrict PortaBilling controlled headers attribute applies within a service policy assigned to a connection, an account (directly or via the product), or within a call authorization rule.

This attribute defines how the c-line (“c=*” line that contains connection information) should be handled in an outgoing SDP message depending on the requirements of different User Agents. You can select one of the following options from the drop-down list:

• 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.

The SDP c-line mode attribute applies within a service policy assigned to a connection or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling or the called party that receives SDP messages from PortaSwitch.

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

This attribute allows to remove ptime from SDP messages in case if legacy IP devices can’t process this attribute.

By default, when this attribute is disabled, ptime isn’t removed from an SDP message.

To enable this attribute, select the checkbox for SDP ptime remove and specify its value:

• No – type in this value to avoid any ptime modifications (the default option).

• All – type in this value (not case-sensitive) to remove all ptime attributes regardless of their value from SDP messages.

• <number> – type in the exact number to specify the duration of media streams in milliseconds so PortaSIP removes ptime if it equals this value.

The SDP ptime remove attribute applies within a service policy assigned to a connection or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling or the called party that receives SDP messages from PortaSwitch.

When this attribute is enabled, PortaSIP registers the first SDP message received in a 1xx SIP response from the callee and sends it to all resulting 1xx SIP responses sent to the caller. This is required in case a caller’s User Agent 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.

To enable the Sticky 1xx SDP attribute, select the checkbox and turn on the toggle switch.

The Sticky 1xx SDP attribute applies within a service policy assigned to a connection or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute is applied to the calling party and is ignored for the called party.

Secure Telephony Identity Revisited (STIR) standard is implemented as a digital signature so that providers can authenticate all the calls they originate. A signature is included in the SIP Identity header and sent to a vendor.

Enable this attribute when configuring the outgoing call authentication to implement STIR/SHAKEN.

To enable the Stir signature required attribute, select the checkbox and turn on the toggle switch.

This attribute should be enabled for the vendors which expect to receive INVITE requests from the PortaSIP with STIR-signed Identity header.

With this attribute, you can specify the list of SIP response codes to stop call forking when a user is either busy or declines the call. When PortaSIP receives these codes, it stops the attempts to deliver an incoming call to other devices of the user.

To add stop call forking codes to the list:

• select the Stop call forking code checkbox;

• specify a code;

• to add the next code to the list, click Add .

To delete a stop call forking code from the list, hover over the code and click Delete .

The recommended codes are:

• 486 - Busy Here

• 603 - Declined

Note that this attribute applies only if the call forking mode specified on the Configuration server is “grouped” or “one-by-one” (on the Configuration server open ClusterSuite > PortaSIP Cluster > select a SIP cluster > MUB2bua group > select the Call forking mode).

The Stop call forking code attribute applies within a service policy assigned to an account to fine-tune call forking for a specific user.

This attribute allows to specify the header 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.

The Transfer caller CLI header attribute applies within a service policy assigned to a connection, an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute applies to the calling party initiating a blind transfer.

By default, when this attribute is disabled, if a blind transfer is unsuccessful (e.g., party A calls party B, B transfers the call to party C and C doesn’t answer), PortaSIP restores the call between A and B.

When this attribute is enabled, it instructs PortaSIP not to reconnect the transferor (party B) back to the transferee (party A) in case of an unsuccessful blind transfer, so the call finishes. This attribute applies to the transferor. It may be useful when calls are transferred by an auto attendant.

To enable the Transfer disable recovery attribute, select the checkbox and turn on the toggle switch.

The Transfer disable recovery attribute applies within a service policy assigned to an account (directly or via the product) or within a dynamically matched service policy. Make sure that the SIP end-point pattern field isn’t empty for the dynamically matched service policy.

This attribute triggers ringback tone generation on 18x SIP signaling during blind transfer. This attribute applies to the party initiating the transfer.

To enable the Transfer progress attribute, select the checkbox and select one of the following options from the drop-down list:

• No indication – PortaSwitch provides no audio indication during the blind transfer.

• Transferor MOH – PortaSwitch plays the Music on Hold (MOH) prompt when the blind transfer is initiated. Early media from the transfer target isn’t relayed.

• Transferor MOH or system – PortaSwitch plays the Music on Hold (MOH) prompt or the default system prompt (if the MOH prompt isn’t selected) when a blind transfer is initiated. Early media from the transfer target isn’t relayed.

• Ringing audio – 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 aren’t received within a predefined timeout;

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

The Transfer progress attribute applies within a service policy assigned to an account (directly or via the product).