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Setting up a SIP Trunk

Turn on the slide to configure a SIP trunk.

Step 1: Choose a service name from the drop-down list. The service names added in the Direct Inward Dial (DID) Numbers page are listed in the drop-down.

Step 2: Choose a preconfigured provider template based on your SIP service provider. If your SIP service provider is not in the drop-down list, choose “Custom”.

Custom

Choose the type of interface connectivity for SIP trunk from the “Use Secondary Interface for Trunk” drop-down list. Primary interface refers to GE 0/0/0 and Secondary interface refers to GE 0/0/1. Primary interface is always connected to the internet service provider. If the internet service provider and SIP trunk service provider are different, use secondary interface for SIP trunk connectivity.

Field Name	Option	Description
Use Secondary Interface for Trunk?	No	SIP trunk and internet connectivity is provided by the same service provider.
	Yes-with static address	SIP trunk and internet are provided by two separate service providers. Internet service provider is connected using the primary interface and SIP trunk service provider is connected using the secondary interface. The SIP trunk service provider provides static IP address for the network connectivity.
	Yes-with dynamic address	SIP trunk and internet are provided by two separate service providers. Internet service provider is connected using the primary interface and SIP trunk service provider is connected using the secondary interface. The SIP trunk service provider provides dynamic IP address for the network connectivity.

Optional:

• In the “External Public Address” field, enter the external public IP address assigned by your internet service provider so that SIP services work across Network Address Translation (NAT).

Option 3 – Secondary Interface with Dynamic Address

Interface Settings

You can opt to leave the fields with the default values. If required, you can modify the optional fields.

Optional:

• • Ethernet ports usually use the auto-negotiate protocol settings. If your switch does not support this option by itself, choose from the following from the “Interface Options” field:
    • Auto Negotiate
    • Gigabit Ethernet
    • Fast Ethernet Full Duplex
    • Fast Ethernet Hakf Duplex
    • Ethernet Full Duplex
    • Ethernet Half Duplex

By default, Auto Negotiate is selected.

• Enter the IP address of the dedicated, private DNS used by your service provider in the “Name Servers” field. Ensure that you enter the name server addresses even if they are provided via DHCP. You can enter a maximum of 6 IP addresses separated by spaces.
• In the “External Public Address” field, enter the external public IP address assigned by your internet service provider so that SIP services work across Network Address Translation (NAT).

Proxy Server

Enter an IP address, Fully Qualified Domain Name (FQDN), or SRV record for proxy address in the “Proxy Address” field.

Optional

• Enter an IP Address, fully qualified domain name, or domain SRV for your service outbound proxy (if one is used), in the “Outbound Proxy Address” field.
• If you have provided an IP address for the outbound proxy, you may also specify a non-standard SIP port if necessary in the “Outbound Proxy Port” field. Leave blank to use port 5060.
• Enter the username and password if your service provider requires authentication for every call in the “Username” and “Password” fields.
• Enter the authentication realm for call authentication in the “Authentication Realm” field. Typically, authentication realm is the service domain name
• Check the “Include in Invite” check box, if your service provider requires authentication details to be sent in the initial invite. If unchecked, authentication is provided in the response to a 407 challenge.

Proxy Server – Advances Options

• • In the “Min-SE” field, enter the minimum value for the session expiry parameter sent in the initial invite. Range is from 90 to 86,400 seconds. Unless instructed by your SIP service provider, the default value of 90 seconds must be used.
  • In the “Session Expires” field, enter the maximum duration of a session in seconds. During a call, the session expiry time is periodically refreshed based on the value entered here. Range is from 90 to 86,400 seconds. Unless instructed by your provider, the default value of 1800 seconds should be used.
  • Limit the range of ports used for RTP. Enter even numbers between 8,000 to 48,198 in the “RTP Port Range” fields.
  • Choose the protocol used for transport layer by your service provider from the “Transport layer” drop-down list.
  • Choose one of the ITU-T T.38 standard Fax Transmission Protocol to be used for a specific VoIP dial peer from the “Fax Transmission Protocol” drop-down list. Available options are:
    • T.38
    • T.38 fall back to G.711 u-law
    • T.38 fall back to G.711 a-law
    • Pass Through G711u
    • Pass Through G711a

By default, the fax transmission protocol chosen is “T38”.

• • Choose one of the following as the DTMF signaling mechanism (based on the protocol offered by your SIP service provider) from the “DTMP Signaling Protocol” drop-down list.
    • RFC2833
    • sip-notify
  • Choose one of the calling party header selection types from the “Calling Party Header Selection” drop-down list:
    • From
    • Remote Party ID (RPID)
    • P-AID Pilot DID
    • P-AID Assigned DIDs
  • Leave the “Calling Party Domain” field blank to send the BE4000 IP address with calling party headers. Enter a domain name or full qualified domain if you want to replace the BE4000 IP address.
  • Choose the “Pilot Number” from the drop-down list if the service provider requires a specific number to be used for P-Asserted Identity Headers.

NOTE: “Pilot Number” field is displayed only when “Calling Party Header Selection” drop-down is chosen as “P-AID Pilot DID”

• Enter the dialing prefix in the “CLI Restriction Prefix” field if the service provider allows calling line ID to be withheld on a call by call basis.
• Uncheck the “RFC3555 Compliant G.729 Annex B” check box if the call server is not RFC3555 compliant for G.729 Annex B SDP formatting (Adds g729-annexb override). Check if unsure.
• Check the “Two way media override” check box to override modification of media stream from send/receive to sendonly or inactive. When checked, two way media is always be requested.
• Check the “Redirection (Option)” to reset the default processing of 3xx messages. By default, SIP gateways process all incoming 3xx redirect messages according to RFC 2543. However, if the Redirection option is disabled, the gateway treats the incoming 3xx responses as 4xx error class responses. Redirection should be selected by default and only unselected if required by the SIP trunk provider.
• Enable “Options Ping” toggle to monitor the SIP service availability allowing traffic to be rerouted, if possible, in the event of failure.
  • Enter the period between Options packets being sent while the service is considered to be up in the “Service Up Interval” field. Range is from 5 to 1,200 seconds. Default is 60 seconds.
  • Enter the period between Options packets being sent while the service is considered to be down in the “Service Down Interval” field. Range is from 5 to 1,200 seconds. Default is 30 seconds
  • Enter the number of missed responses allowed before a service is considered unavailable in the “Retries” field. Range is from 1 to 10. Default is 5.

Registrar

Registrar server can be configured either through DHCP or by providing IP address and port. Click one of the following options based on your network:

• Configure via DHCP
• Configure address and port

Configure via DHCP

 

Step 1: Enter the authentication realm used for registration by your service provider in the “Authentication Realm” field.

Step 2: Enter the username and password, if the service provider requires per call authentication in the “Username and Password” fields.

Step 3: If the service provider requires a DID to be included with registration authentication, choose the appropriate DID number for each Username and Password. Else, choose “No”.

Optional:

• Check the “Registrar with Realm” configuring the registrar with the realm information provided for the proxy. Uncheck to remove the configuration.
• Click “Add Row” to add multiple rows for username, password, and include DID. You can add a maximum of 12 rows.
• Enter the registration timeout period in the “Registration Timeout” field. Determines how frequently the system registers.
• Choose TCP or UDP from the “Transport Layer” drop-down list as the protocol used by the service provider for Transport Layer.

Configure Address and Port

Optional:

• Enter an IP Address, fully qualified domain name, or domain SRV for service registrar in the “Registrar Address” field.
• If you have provided an IP address for the registrar, you may also specify a non-standard SIP port if necessary for registrar port in the “Registrar Port” field. Leave blank to use port 5060.
• Enter the authentication realm used for registration by your service provider in the “Authentication Realm” field.
• Check the “Registrar with Realm” configuring the registrar with the realm information provided for the proxy. Uncheck to remove the configuration.
• Enter the username and password, if the service provider requires per call authentication in the “Username and Password” fields.
• If the service provider requires a DID to be included with registration authentication, choose the appropriate DID number for each Username and Password. Else, choose “No”
• Click “Add Row” to add multiple rows for username, password, and include DID. You can add a maximum of 12 rows.
• Enter the registration timeout period in the “Registration Timeout” field. Determines how frequently the system registers.
• Choose TCP or UDP from the “Transport Layer” drop-down list as the protocol used by the service provider for Transport Layer.

Security

You must add at least one trusted IP Address. The BE4000 accepts incoming VoIP (SIP) calls only if the remote IP address of an incoming VoIP call matches an address in the Trusted IP Address list. Enter the IP addresses provided for proxy, outbound proxy, and registrar from your service provider. IP addresses must be provided if hostnames are used. Entries can be provided either as a host address (x.x.x.x) or subnet (x.x.x.x /nn).

Click “Add Row” and enter the trusted IP addresses.