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The SIP Profile Advanced Settings object is an extension of the SIP Profile object. When communicating with an external gateway there are various attributes/features that need to be associated with that particular gateway. The SIP Advanced Settings object is created under the SIP Profile object and adds more attributes/features to the existing SIP Profile object. Once configured, the SIP Profile created can be added to the SIP Signaling object, the External SIP Gateways object and External ENUM Server Sets object. The SIP Profiles configured will all be selectable from a drop down menu in each of the objects described above. In addition to the fields in the SIP Profile and SIP Advanced Setting object, there are also multiple additional objects that can be configured under the SIP Profile (SGP). These additional objects add additional features/attributes. Refer to the information below for information on configuring each individual field.

Web GUI Page

Dialogic > Profiles > SIP Profiles > SIP Profile > New SIP Advanced Settings

 

Maximum Objects

One SIP Advanced Settings object per SIP Profile object.

Related Topics and Dependencies

The SIP Advanced Settings object is a extension of the SIP Profile object. The SIP Advanced Settings object adds attributes/features to the already created SIP Profile.

SIP Profile - SGP 

Field Descriptions

Outbound Modem Triggers Re-INVITE

Disabled (Default) - Feature is disabled.

Enabled - The IMG 2020 will send a re-INVITE to the far-end when it detects modem traffic. The IMG 2020 switches the RTP into Modem Bypass mode over G.711. The IMG 2020 will then use the bypass codec type specified in the associated Bearer Profile (u-law or a-law). Incoming Re-INVITE's during modem calls are accepted automatically.

INFO for Spirou/ITX

The ITX/TXA messages are used in a Spirou (Signalisation Pour l'Interconnexion des Réseaux Ouverts/Signaling for the Interconnection of Open Networks) network for audio services that are paid for by either a flat rate calculation or time based calculation. The IMG 2020 supports sending both the ITX and TXA messages. The INFO for Spirou/ITX field enables or disables the sending of these messages. Refer to SPIROU/ITX in SIP INFO topic for more information.

Disabled (Default) - Support for the Spirou Variant is disabled.

Enabled - When enabled there are two scenarios. SS7 to SIP and SIP to SS7. See below.

  • SS7 to SIP - The IMG 2020 will accept the ITX message from the SS7 ISUP leg and interwork it into a SIP INFO Message on the SIP leg.
  • SIP to SS7 - The IMG 2020 will accept the SIP INFO message (RFC 2976 - The SIP INFO Method) from SIP leg and generate an ITX message onto the SS7 ISUP leg. 

Outgoing Fully Qualified Domain Name (FQDN)

The Fully Qualified Domain Name is a domain name which specifies an exact location of a piece of equipment within a Domain Name System. (Ex. myhost.example.com). The IMG 2020 allows user to input a FQDN in a number of different objects. The FQDN is entered in place of the IP address. If the Outgoing Fully Qualified Domain Name field is enabled, the IMG 2020 will include the FQDN in its outgoing requests and response messages. The feature is disabled by default and can be modified to send out the FQDN in place of IP addresses for the local SIP Signaling port, the local VoIP Module Port, or both. See below.

Disabled (Default) - Feature is disabled. Only IP addresses are sent with outgoing requests and responses.

Signaling Only - The Fully Qualified Domain Name assigned to the local SIP Signaling port in the SIP Signaling object is inserted in the outgoing SIP messages. See topic.

SDP c=line Only - The Fully Qualified Domain Name assigned in the VoIP Resource object is inserted in the outgoing SIP messages and responses.

Both - The Fully Qualified Domain Name assigned to both the VoIP Resource and SIP Signaling objects are inserted in the outgoing SIP messages.

User Domain - The Fully Qualified Domain Name assigned to the User Domain. The outgoing SIP INVITE message To: and From: headers to contain the FQDN in SIP IP address.

Incoming Reason Header Cause Code

The Reason Header describes through the use of cause codes why a SIP request or response was issued. The Reason Header is issued in the BYE, CANCEL, 4xx, 5xx, and 6xx requests/responses. The Reason Header field provides the ability to propagate cause code information from SIP leg to TDM leg and/or TDM leg to SIP leg without having to configure SIP-T. The SIP Reason Header will provide additional information on why a call was disconnected. The cause codes implemented are based on ITU-T Recommendation Q.850. The interworking of cause code values are based on RFC 3326.

Disabled (Default) - The IMG 2020 will interwork of the release cause codes based on RFC 3398 ISUP to SIP Mapping.

Enabled - The IMG 2020 will propagate the cause value from the first Reason Header with a valid cause value based on ITU-T Recommendation Q.850.

For more information refer to SIP Reason Header.

Allow 180 after 183

In a normal call flow the 183 Session Progress message indicates whether the calling user agent should provide alerting or not. Once the 183 has been sent then a 180 Ringing message is sent dependent on the specific call flow. The Allow 180 after 183 field can be configured either to allow or suppress the 180 Ringing message. See below.

Enabled (Default) - The IMG 2020 will send a 180 Ringing message after sending a 183 Session Progress message if the specific call flow requires.

Disabled - The IMG 2020 will not send a 180 Ringing message if a 183 Session Progress with SDP has already been sent to the remote SIP side.

Acceptable Inbound Call Type

The Acceptable Inbound Call Type field is for inbound calls only and is applicable if an external gateway communicating with IMG 2020 is configured to use TLS. The field is used to specify whether non-TLS inbound calls will be accepted or rejected by the IMG 2020 from this gateway. (i.e. UDP or TCP call attempts) The Acceptable Inbound Call Type field enforces the use of TLS as a preference or a requirement for a specific SIP profile assigned to an external gateway. See below.

Make encrypted calls-accept all calls (Default) -  If a non-encrypted call is received from an external gateway, the IMG 2020 server port will accept the call.

Make and accept only encrypted calls - If a non-encrypted call is received from an external gateway, the IMG 2020 server port will not accept the call.

Secure SIP (SIPS)

SIP provides a secure URI called SIPS URI. If an external gateway and the IMG 2020 are configured to use TLS security, then this field is used to select the URI scheme sips or sip. If TLS is not configured on the IMG 2020 then the Enable SIPS field will be ignored. Refer to the Configure TLS topic for information on configuring TLS functionality. If TLS is configured on IMG 2020 then the selections below are valid. See example below.

Enabled (Default) - When set to true the SIP URI scheme is formatted as follows:

sips:alice@atlanta.com;transport=tcp

Disabled - When set to false the SIP URI scheme is formatted as shown below:

sip:alice@atlanta.com;transport=tcp

SIP-T Pass On

When SIP-T is enabled on the IMG 2020, it is enabled globally. If however there is an external gateway that SIP-T is either not supported on or not enabled, the IMG 2020 can disable SIP-T to that gateway. In the SIP-T Pass On field there is a drop down menu where you can select yes or no.

Enabled - If Yes is selected, SIP-T will be enabled and any ISUP messaging will be sent between SIP and SS7.

Disabled - If No is selected, SIP-T will be disabled and no SIP-T messaging will be sent between SIP and SS7.

If no is selected and the external gateway sends the IMG 2020 SIP-T messaging, the IMG 2020 will not transfer the SIP-T ISUP information. The IMG 2020 will operate as if SIP-T were not enabled and pass only the SIP messaging.

Hold/UnHold

Use this field to configure the IMG 2020 to On Hold/Resume (Hold/UnHold) while in early media as well as in connected state.

Accept (Default) - IMG 2020 will accept the Hold/UnHold request.

Reject - IMG 2020 will reject the Hold/UnHold request with SIP 488 "Not Acceptable Here" 

Hold/UnHold Propagation

Use this field to configure the IMG 2020 to either propagate or not propagate a On Hold/Resume (Hold/UnHold) request to the outbound SIP. The Hold/UnHold is supported in early media (SIP UPDATE) as well as in connected state (SIP UPDATE or SIP RE-INVITE). See below.

The Call Flows below display a SIP to SIP call. The Hold/UnHold Propagation feature supports SIP to SS7 and SS7 to SIP as well.

Enabled (Default) - When the Hold/UnHold Propagation field is set toenabledthen the SIP UPDATE or SIP INVITE that contains the Hold message will be propagated to the remote leg of the call.

Disabled - When the Hold/UnHold Propagation field is set todisabledthen the SIP UPDATE or SIP INVITE that contains the Hold message WILL NOT be propagated to the remote leg of the call.

Retry After Support

The IMG 2020 supports sending and receiving the Retry Header in the event there is SIP congestion. When the Retry after support field is set to enable, the IMG 2020 would determine the level of SIP congestion. If it is determined that SIP congestion levels are acceptable, the call will go through. If however congestion levels are high, the IMG 2020 will send a 503 Service Unavailable along with a Retry Header indicating time to wait before re-sending INVITE back to IMG 2020. Refer to SIP Retry-After Header - Transmit and SIP Retry-After Header - Receive for more information.

Enabled (Default) - The IMG 2020 will respond with a Retry-After Header in its response if the Channel Group is congested.

Disabled - The IMG 2020 will not send a Retry-After Header in its response if Channel Group is congested.

Clear Channel Override

The IMG 2020 supports the Clear Channel Codec as described in RFC 4040.

Disabled (Default) - Clear Channel Override is disabled. The Outgoing Codec will not be forced to Clear Channel even if the incoming parameters dictate the correct conditions. See IMG 1010 - Clear Channel Codec for more information.

Enabled - If the Bearer Capability value of the incoming call matches a certain set of requirements and the Clear Channel Override field in the SIP SGP Profile pane on the outgoing side is set to enable then the IMG 2020 will force the outgoing Codec to be clear channel. See Clear Channel Codec for more information.

Phone Context

The Phone-Context Parameter (RFC 3966) is supported on the IMG 2020 for outgoing INVITE requests only. Select from drop down menu one of the following choices. See SIP Phone Context Parameter for more information.

Disabled (Default) - Phone Context support is by default disabled.

Include in To: Header - Include the Phone Context Parameter in the To: Header only.

Include in From Header - Include the Phone Context Parameter in the From: Header only.

Include in both To: and From: Headers - Include the Phone Context Parameter in both the To: and From: Headers.

Phone Context String

If Phone Context Support field above is set to Disable then the Phone Context String field cannot be modified. To be able to modify the Phone Context String field, change the Phone Context Support field to one of the choices other than Disabled. Once accomplished, click in the Phone Context String field and enter the Phone Context String which can be one of two formats. The two formats are either by digit string or by domain name. Max number of characters is 64. Below are a few examples:

Digit String = 1508862

Domain Name = hyannis.dialogic.com

Allow NOTIFY w/o Subscription

Choose from a drop down menu the following selections:

Disallow (Default) - Do not allow the IMG 2020 to accept a NOTIFY message without a subscription.

Allow - Allow the IMG 2020 to accept a NOTIFY message without a Subscription. The Notify message will be accepted without a subscription only if the parameters Content Type = application/simple-message-summary and Event = message summary are in the SIP NOTIFY message. Select Allow when configuring the Message Waiting Indicator feature.

183 Periodic Retransmission

In response to a SIP INVITE message, a 183 Session Progress message is first received on the B Leg. It is interworked and transmitted out the A leg. If the Call Answer Timeout value in the field below is set to a large amount of time, the IMG 2020 will keep retransmitting the interworked 183 Session Progress messages. Typically one 183 Session Progress message every minute. The 183 Periodic Transmission field can either enable or disable the retransmission of these 183 Session Progress messages out the A Leg. Refer to the Call Answer Timeout and 183 Periodic Retransmission functionality topic for more information.

Enabled (Default) - If 183 Session Progress message is interworked from the B Leg to the A leg the IMG 2020 will retransmit the 183 Session Progress message out the A Leg as long as the call has not been answered (200 OK received).

Disabled - If 183 Session Progress message is interworked from the B Leg to the A leg the IMG 2020 will NOT retransmit the 183 Session Progress message out the A Leg as long as the call has not been answered (200 OK received). However, the IMG 2020 still sends out the retransmitted 183 Session Progress messages interworked from the B leg.

Star Digit Mapping

This will allow the ability to handle star () as digit number. For example, this mapping of the star digit () is required for some JT-ISUP use cases.

Mapped to 0x0E (Default) - The digit is mapped to digit 0x0E.

Mapped to 0x0B - The star digit (*) from SIP is mapped to digit 0x0B.

To receive and send digits A to F, the Support Digit A to F field in the must me set to True.

Route by Prefix

This will allow the user to associate a Prefix with an outgoing Channel Group.

Disabled (Default) - Route by Prefix is disabled. 

Enabled - Route by Prefix is enabled.

Enabled - Keep Prefix - Route by Prefix is enabled and the Prefix is relayed to the outgoing leg.  This is applicable for SIP-SIP call only.


Call Answer Timeout

After a 183 Call Session Progress message is received, a Call Answer Timeout timer starts. If this timer expires, the call is cancelled. During a SIP Early Media call, the Call Answer Timeout may need to be extended to accept the early media. The Call Answer Timeout field allows a user the ability to modify the Timeout period. The default value is 3 minutes and can be modified to not expire for up to 60 minutes. For more information on the functionality of the Call Answer Timeout field, refer to the Call Answer Timeout and 183 Periodic Retransmission functionality topic.

200OK PRACK/UPDATE Race Condition Prevention Delay

This field allows a user the ability to introduce a delay to prevent a race condition between the 200 (OK) for the INVITE and the 200 (OK) for the PRACK/UPDATE. This condition is sometimes encountered when communicating with some Voice over LTE (VoLTE) implementations.

The minimum value is 0x100msec = 0sec. The maximum value is 20x100msec =  2sec. If the user selects between 0-20, this would be 100msec increment. When the delay is set to 0sec, the feature is disabled. There is no delay added, which is the default behavior.

Append (+) for Headers

Important: The next two fields (Append + for Headers and Remove + for Headers) apply to the Pass through + character in the user part of URI feature. The prefix + cannot be used for routing. The feature simply adds, passes through, or removes the + prefix.

Use the Append () for Headers field to add the prefix + to any outgoing INVITE requests if the incoming INVITE does not have +. Click on one or more selections within the Append () for Headers field. The chosen selections (highlighted in blue) will have the prefix + appended to the outgoing INVITE. To highlight more than one choice, hold down the <Ctrl> key while clicking the selection.

Remove (+) for Headers

Use the Remove () for Headers field to remove the + prefix from an incoming message. The outgoing INVITE request will not include the + prefix. This feature can also be used in the case where the incoming side is not SIP. Click on one or more choices within the Remove () for Headers field. The chosen selections (highlighted in blue) will have the prefix + removed from the outgoing INVITE. To highlight more than one choice, hold down the <Ctrl> key while clicking the selection.

R-URI Header Tags

Click on the selection within the R-URI Header Tags field. The tag selected (highlighted in blue) will be added into the R-URI of the outgoing INVITE in the sip_uri_enc method. To highlight more than one selection, hold down the <Ctrl> key while clicking the selection. Selections are described below.

None (Default) - No tags will be added to outgoing INVITE.

CIC (Carrier Identification Code)and DAI (Dial Around Indicator) - The CIC parameter is a three or four digit code used in routing tables to identify the network that serves the remote user when a call is routed over many different networks. The DAI parameter will be displayed immediately after the CIC code. See SIP CIC and DAI Codes for more information.

RN (Routing Number) - The RN tag is used to convey the location routing number. Refer to the Local Number Portability (LNP) topic for more information.

NPDI (Number Portability DIP Indicator) - The NPDI tag is used to indicate whether an LNP query has been performed. Refer to the LNP Routing topic for more information.

ISUB/ISUB-Encoding - The ISDN calling party Subaddress parameters convey address information of each device in the ISDN network. For more information on ISUB Encoding, refer to the ISUB Encoding topic. 

IP Traffic Management Response Message

This field allows a user the ability to select the SIP response message when IP Traffic Management rejects the call.

503 Service Unavailable (Default) - The server is currently unable to handle the request due to a temporary overloading or maintenance.

500 Server Internal Error - The server encountered an unexpected condition which prevented it from fulfilling the request.

403 Forbidden - The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated.

 

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