Creating a local VoIP trunk

Create a VoIP trunk for inbound and outbound calls at the branch office.

The office peer trunk setup is only required if you use the following gateways in the branch office:

• GO04, GO08

• GT01, GT02

To create a VoIP trunk, you must meet the following requirements:

Prerequisite

• In the Deployment, complete tasks 1 to 11.

• Select Partially managed for Management Mode when adding a survivability branch. For details, see Adding a survivability branch.

• Select Handled by branch for Local call handling. For details, see Adding a survivability branch.

Procedure steps

1. Connect to the GUI of the FortiVoice LSG unit.
2. Go to Trunk > VoIP > SIP.
3. Click New.
4. Configure the following:

   GUI field	Description
   Enabled	Select to activate the SIP trunk.
   Name	Enter a name for this trunk such as the name of the VoIP service provider.
   Display name	Enter your caller ID that will appear on the called phone, such as Example Company. For details about the caller ID hierarchy, see the Caller ID modification section in the FortiVoice Cookbook.
   Main number	Enter the phone number provided by the VoIP service provider.
   SIP Setting
   SIP server	Enter the VoIP provider’s IP address or domain name. For example, 172.20.120.11 or voip.example.com.
   SIP port	Most SIP configurations use TCP or UDP port 5060 for SIP sessions. If your VoIP service provider uses a different port for SIP sessions, enter the port number. If you select the Using SRV record option, this field is greyed out.
   Using SRV record	If you entered the VoIP provider’s domain name in the SIP server field, select this option to translate the domain name and obtain the SIP port. You can only select this option if your VoIP provider uses the same setting.
   User name	Enter the user name provided by the VoIP service provider for the FortiVoice LSG unit to register with the SIP server.
   Password	Enter the password provided by the VoIP service provider for the FortiVoice LSG unit to register with the SIP server.
   Auth. user name	Some VoIP providers may provide you with an authentication user name that is different from your user name for the FortiVoice LSG unit to register with the SIP server. If that is the case, enter the authentication user name here.
   Realm/Domain	Some VoIP service providers’ SIP servers authenticate the PBXes that register with them by requesting the name of the host performing the authentication. If this is the case with your VoIP service provider, enter the name of the host performing the authentication provided by your VoIP service provider.
   SIP settings	Select the SIP profile to apply the supported phone features and codecs for the trunk. To match the information of the VoIP service provider, you can edit the existing profile or click + to add a new one.
   Max channel	Each trunk contains multiple channels. The number of channels you can have in a trunk is controlled by your VoIP service provider. This number displays under line appearance option when you configure programmable phone keys for phone profiles. Consult your VoIP service provider for the maximum of channels that you can set to limit the number of concurrent calls. For example, if you want to allow six calls at a time, enter 6. The value range is from 1 to 450.
   Overflow check	If selected, the phone calls exceeding the Max channel limit will be handled according to the call handling actions set in the dialplan applied to this trunk. If unselected, the phone calls exceeding the Max channel limit will be disconnected.
   Max outgoing channel	With known max channels, if you need to reserve incoming channels, you may enter the number of outgoing channels allowed and the remaining channels are for incoming calls. For example, if the max channel number is 10 and you want to reserve 4 channels for incoming calls, then you can enter 6 for Max outgoing channel. The value range is from 0 to 2000.
   User=Phone in SIP URI	Select if your service provider requires this option to make the FortiVoice LSG unit to be compatible with the VoIP service provider’s configurations.
   Inband ringtone (Early media)	Select to enable the FortiVoice LSG unit to send ring tone to the caller of an incoming call before the establishment of a call connection.
   Caller ID Option	Configure how to display your caller ID to meet the different requirements and scenarios of the service providers. Caller ID information is contained in the From header and P-Asserted-Identity header of SIP packets sent to the called phone's PBX. From header: The From header field indicates the identity of the initiator of the call request from the point of view of the PBX server. SIP user name: Select if you want the user name provided by the VoIP service provider for the FortiVoice LSG unit to register with the SIP server to appear on the called phone. See User name. Caller ID priority rule: Select if you want to configure your FortiVoice caller ID according to the FortiVoice caller ID priority hierarchy. For more information, see the FortiVoice Phone System Administration Guide. Main number: Select if you want the trunk main number to appear on the called phone. See Main number. Specified: Enter the ID you want to display on the called phone in the format of display name <number>. P-Asserted-Identity header: This header contains the caller ID information for the call on the INVITE SIP packet. No PAI header: Select if you want to disable PAI header. Caller ID priority rule: Select if you want to configure your FortiVoice LSG unit's caller ID according to the FortiVoice caller ID priority hierarchy. For more information, see the FortiVoice Phone System Administration Guide. Main number: Select if you want the trunk main number to appear on the called phone. See Main number. Specified: Enter the ID you want to display on the called phone in the format of display name <number>. Diversion Header Use: Allows you to set the use of the diversion header for call twinning, call forwarding, or normal outbound calls. On Redirect: Use the diversion header during call twinning and call forwarding. No Diversion Header: Disable the use of the diversion header for all calls. Always: Allow the diversion header to be applied to all calls. Diversion Header Source: DID Mapping: Use the mapped DID if available, then use the caller ID (CID) of the main trunk. Trunk Caller ID: Use the caller ID of the main trunk.
   Registration
   Type	Enter the SIP registration information from the VoIP service provider by selecting a registration method in Type. You can receive calls after registering with the SIP server of the VoIP service provider. Disable: Select to deactivate the registration with the VoIP service provider. Standard: Select to use the standard registration method which automatically registers with the SIP server of the VoIP service provider. Enter the registration interval in minutes. Registration URI: Enter the registration string provided by the VoIP service provider. The string in Registration URI has the following format: <user>@<host><:port> where <user> is the user name. <host> is a hostname, domain name, FQDN, or IP address. <:port> is the port number. If you omit to specify a port, the default port (5060) is used. Examples: support@mycompany.com support@mycompany.com:6000 bob@168.176.248.255 Registrar: Select to enter the registration information from the VoIP service provider: Registrarhost/IP: Enter the VoIP service provider’s SIP registration server domain name or IP address. For example, 172.20.120.11 or voip.example.com. Registrar port: Most SIP configurations use TCP or UDP port 5060 for SIP sessions. If your VoIP service provider uses a different port for SIP sessions, enter the port number. Transport protocol: Select the transport protocol used for the registration. Registration interval: Enter the registration interval with the SIP server in minutes.
   Outbound Proxy	Some VoIP service providers use proxy servers to direct its traffic. If this is the case, your registration request will go to the proxy server first before reaching the registration server. Configure the following: Select to activate the proxy server setting. Proxy (Host/IP): Enter the proxy server’s domain name or IP address. For example, 172.20.120.11 or voip.example.com. Proxy port: Enter the port number of the proxy server. Transport protocol: Select the transport protocol used for the registration.
   Fax
   Automatic fax detection	Selecting this option may cause the following behaviors: Delay the call response time on this trunk by automatically adding two ring tones to detect incoming fax signals. Affect toll charges on incoming lines. Select for the FortiVoice LSG unit to detect incoming fax signal on this trunk automatically.
   Forward to DID mapping extension	This option is available when you select Automatic fax detection. Select this option if a DID number is mapped directly to an extension to receive voice and fax calls. Faxes will be sent to the extension’s personal fax account, accessible through the FortiVoice User Portal. In Forward to eFax account (next field), select an eFax account. Should a fax fail to be received by the DID mapping extension, the FortiVoice unit will use this eFax account as a fallback.
   Forward to eFax account	This option is available when you select Automatic fax detection. Select or edit an eFax account to receive faxes. To add a new eFax account, click +.
   Phone Number	Adding a phone number in this field is optional and for information purposes only. The phone number supports digits from 0 to 9 and a maximum of 63 digits. Click New to add a phone number provided by your VoIP service provider. Click Create when done. You can add multiple phone numbers.

5. Click Create.
6. You have completed this procedure. You can now go to Adding dial plans.