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2.1 The background of SIP Trunk

The SIP Trunks serve connecting PrivateServer to a local PBX or to a SIP Voice provider. The above statement implies that a Trunk can be used both for receiving and placing calls in a bidirectional way.

The PrivateServer is designed to perform always as a client when configuring a SIP Trunk. This means that PrivateServer itself can not act as a Trunk Provider.

Even if a SIP Trunk can receive and place calls PrivateServer has to be configured to act in a bidirectional way. The Trunk setup should be done for the incoming calls first and after this function is enabled you can setup it for outgoing calls as well.

The incoming calls Trunks are configured in the Inbound entry that you access under Main Menu -> PBX INTEGRATION. Outgoing calls are configured into the Outbound entry that is under the Inbound one.

2.2 Inbound Configuration

 In 

2.2.1 Create an Inbound Trunk

1. username and password: you must insert username and password
2. IP based: leave the username and password fields blank and fill in the HOST parameter.

The fields have to be configured as follows:

• NAME: a meaningful name for this trunk
• HOST: IP address of the SIP server provided by ITSP
• OUTBOUND PROXY: IP address of the outbound proxy server provided by ITSP
• VIRTUAL PHONE NUMBER: associated virtual number
• USERNAME: login username
• PASSWORD: password for the username entered above
• PORT: registration port of the service
• REGISTER: select if you want to have only outgoing calls or incoming (see below for deeper explanation)
• CONCURRENT CALLS: maximum number of calls on a single trunk
• NAT: enable/disable the Network Address Translation configuration
• DIRECTMEDIA: experimental sound management in P2P mode without routing the audio stream through the server
• SENDRPID: the Remote Party ID, used to interconnect with VoIP Provider for the management of privacy
• DTMFMODE: tone signaling technology used for this SIP interconnection
• ALLOW: voice codecs used in this trunk
• DISALLOW: voice codecs denied in this trunk

The default values for Port, Allow and Disallow are usually work well, however feel free to change them to suite your needs better.

Mandatory fields are:

• Host
• Port
• Allow/Disallow

Register option

If the trunk is going to be also used for OUTBOUND calls (external calls) you need to be sure that the "Register" checkbox is checked. In this case the "Name" and "Secret" fields are mandatory as well. Otherwise you can uncheck the "Register" checkbox so the above fields will become optional.

Fill out the fields accordingly to the guidelines above and when you are done click on the "Update" icon at the bottom of the page.

You can get back to the "Sip Trunk List" page where the Sip Trunk table is now filled based on your settings (

A notification will inform you about the progress of the operation. The number shown in the notification is an internal identifier in the database and thus can be ignored. 
To check the entered data of the SIP Trunk click on the "Host" value into the table (for example in this case you should click on "10.0.0.101"). This will open again the Edit Sip Trunk page with the previously entered data.

2.2.2 Change Inbound Trunk settings

On the Edit Sip Trunk page you can change the values of the Inbound Trunk fields. When you are done commit the changes by clicking on the Update icon.
Click again on the "Host" value and verify the values of the fields are actually changed.

2.2.3 Delete an Inbound Trunk

To delete a trunk you need to select it in the trunk list. Click on the selected trunk and you'll be taken to the Edit Sip Trunk page. Click on the Delete icon at the bottom of the page to delete the Sip Trunk.exposed by your peer PBX:

1. SIP Account
2. IP based

4.3.1 SIP Account authentication

If your PBX peer on the other end of the Trunk has given to you a SIP Account to be used for the Authentication, then you need to insert login and password in Username and Password fields. Plus you should enable the Register option (check 

Typically this configuration is used by public SIP providers because they can't rely just on the IP authentication (the customer's PBX could be behind an ADSL, for instance) and they need to be sure you're just the one who has the right to be connected despite the IP address you are using. 

4.3.2 IP Address Authentication

In an enterprise scenarios the authentication is generally based on IP address, because inside enterprise's infrastructure all the structural elements use to have both private and static IP addresses. 

In this case the IP address is most likely an internal one and it should be fixed: these features make it a reliable source for the authentication. 

The IP Address Authentication is based on the Host field in the form of the Trunk creation. 

Plus you need to enable the Send keep alive option (please see 

The actual default value for the keep-alive interval is 60 seconds. You can configure the general keep-alive timeout in the NAT configuration form. Please read PSAM 2.4 Advanced configurations to get informations about it.