Setting up VoIP trunks

You can add one or more VoIP service providers to the FortiVoice unit trunk configuration. The VoIP service providers deliver your telephone services to customers equipped with SIP-based PBX (IP-PBX).

To view the list of VoIP service providers

1. Go to Trunk > VoIP > SIP.

GUI field	Description
Test	Select to test if the trunk is created successfully. The duration of the test call is limited to 60 seconds. For more information, see Testing SIP trunks.
FortiCall	Select to create a SIP trunk with Fortinet’s FortiCall service. You can only create one trunk with FortiCall and use it free for 30 days or 300 minutes, whichever comes first. Note that the trial account only allows outbound calling and no international calling is available. If you sign up for the service during a trial, the trial is closed and billing will start. For more information, see Creating a SIP trunk with FortiCall service.
Enabled	Select to activate this trunk.
Name	The name of the VoIP service provider.
Server	The VoIP provider’s domain name or IP address. For example, 172.20.120.11 or voip.example.com.
Port	The port for SIP sessions.
SIP Setting	The SIP profile applied to this trunk.
Status	The status of the SIP trunk. Not registered: The trunk is not registered with the VoIP service provider and is not in service. In service: The trunk is registered with the VoIP service provider and is in service. Unavailable: The trunk is not reachable. Alarm detected: There is a problem with the phone line. Admin down: The trunk is disabled. Unmonitored: The trunk is unknown.

To create a VoIP trunk

1. Go to Trunk > VoIP > SIP.
2. Click New.
3. Configure the following:

   GUI field		Description
   SIP
   	Enabled	Select to activate the SIP trunk.
   	Name	Enter the name of the VoIP service provider.
   	Display name	Enter your caller ID that will appear on the called phone, such as Example Company.
   	Main number	Enter the phone number that will appear on the called phone.
   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 unit to register with the SIP server.
   	Password	Enter the password provided by the VoIP service provider for the FortiVoice 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 unit to register with the SIP server. If tat 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 New to add a new one. For more information, see Configuring SIP profiles.
   	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. See Configuring 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. This field accepts value in the range of 1-450 inclusive.
   	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. For information on dialplans, see Configuring call routing. 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, the max channel number is 10 and you want to reserve 4 channels for incoming calls, you can enter 6 for Max outgoing channel. The maximum channel limit is 2000.
   	User=Phone in SIP URI	Select if your service provider requires this option to make the FortiVoice unit to be compatible with the VoIP service provider’s configurations.
   	Inband ringtone (Early media)	Select to enable the FortiVoice 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 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: Emergency caller ID: See Setting extension user preferences in Setting up local extensions. External caller ID: See Setting extension user preferences. DID mapping caller number: See Configuring inbound dial plans. Trunk caller ID: See Main number. PBX caller ID (Main display name): See Configuring phone system settings. 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 caller ID according to the FortiVoice caller ID priority hierarchy: Emergency caller ID: See Setting extension user preferences in Setting up local extensions. External caller ID: See Setting extension user preferences in Setting up local extensions. DID mapping caller number: See Configuring inbound dial plans. Trunk caller ID: See Main number. PBX caller ID (Main display number): See Configuring phone system settings. 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	Enter the SIP registration information from the VoIP service provider by selecting a registration method in Type field. 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 in the Registration URI field. The string usually has the following formats: register => user[:secret[:authuser]]@host [:port][/extension] or register => fromuser@fromdomain:secret@host or register => fromuser@fromdomain:secret: authuser@host:port/extension For example, a string could be: register => 2345:password@mysipprovider.com/1234 Registrar: Select to enter the registration information from the VoIP service provider: Registrar host/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 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 (see details in To map a personal fax DID number). Faxes will be sent to the extension’s personal fax account, accessible through the User Portal. In Forward to eFax account (next field), select an eFax account (as configured in Receiving faxes). Should a fax fail to be received by the DID mapping extension, the FortiVoice will use this eFax account as a fallback.
   	Forward to eFax account	This option is available when you select Automatic fax detection. Select an eFax account to receive faxes. To configure an eFax account, see Receiving faxes.
   Phone Number		This is optional. Incoming calls to other DIDs can go through as long as they are forwarded from your VoIP service provider with the correct header information.

4. Click Create.

Testing SIP trunks

After you create a SIP trunk, you can select the trunk and click Test to see if the trunk works.

For more information, see Test.

To test a SIP trunk

1. Go to Trunk > VoIP > SIP.
2. Select the trunk that you want to test and click Test.
3. Select Test Call-Dry Run if you want to run a system SIP trunk test without making a real phone call, or Test Call if you want to test the SIP trunk by making a real phone call.

   The System Configuration Test page appears.

4. Configure the following:

GUI field		Description
Test Call - Dry Run		Run a system SIP trunk test without making a real phone call.
	Destination number	Enter a destination number to call.
	From number	Enter the number from which you want to call the destination number. The FortiVoice unit will connect this number with the destination number for the test.
	Test	Click to start the dry run test and check the Test result.
	Reset	Click to remove the test result in order to start a new test.
Test Call		Test the SIP trunk by making a real phone call.
	Destination number	Enter a destination number to call.
	After call is established	Select the FortiVoice action once it calls the destination number: Play welcome message: The FortiVoice unit will play a message to the destination number. Connect test call to number: In the Number field, enter the number from which you want to call the destination number. The FortiVoice unit will connect this number with the destination number to test the trunk.
	Test	Click to start the test and check the Test result.
	Reset	Click to remove the test result in order to start a new test.

Creating a SIP trunk with FortiCall service

You can create one trunk with FortiCall and use it free for 30 days or 300 minutes, whichever comes first. Note that the trial account only allows outbound calling and no international calling is available.

If you sign up for the service during a trial use, the trial is closed and billing will start.

To create a SIP trunk with FortiCall service

1. Go to Trunk > VoIP > SIP.
2. Click FortiCall.

   The Create SIP Trunk dialog box displays.

3. Note down the MAC Address and System ID for use if you decide to sign up for the service later.
4. Keep Create dialplans for this trunk selected unless you want to create the dialplans by yourself.

   The auto-generated dialplans will replace the default inbound, outbound, and emergency call dialplans. You can delete them if you do not choose to use the FortiCall service.

5. Click OK.
6. For Fax, see Fax.
7. For Register Trial Account, enter your name, email address, and reseller or partner code.
8. Click Create.
9. Click OK.

   The FortiCall trunk is created. You will receive an email with sign up and login instructions.