Handle Outbound Calls Through SIP Trunk

This topic describes how PortSIP PBX handles outbound calls through a SIP trunk. Specifically, the INVITE message structure and how the SIP field values are populated.

DID Pool

To understand the concept of Direct Inward Dialing (DID), please refer to the article What is Direct Inward Dialing (DID)?

Since PortSIP PBX is a multi-tenant system, if multiple tenants set up trunks from the same provider and use the same DID number for their inbound rules, the PBX would not know which tenant to route the incoming call to. Similarly, if an extension from one tenant uses an outbound caller ID that belongs to another tenant, it could cause conflicts.

To prevent these issues, PortSIP PBX introduced the concept of a DID pool, which is a designated range of DID numbers assigned to each tenant.

Trunks Added by System Admin

When a System Admin assigns a trunk to a tenant, they must configure a unique DID pool for that tenant. The DID pool cannot overlap with those assigned to other tenants. Once the trunk is assigned, the tenant can only use DID numbers from their specific DID pool when creating inbound rules.

Trunks Added by Tenant Admin

If a Tenant Admin adds a trunk, they are required to specify a DID pool for that trunk. Any inbound rules created by the tenant must use DID numbers from this pool. As with trunks added by the System Admin, the DID pool must be unique and cannot overlap with DID pools from the same trunk provider.

For example, if Tenant A configures a trunk with provider XYZ and sets a DID pool of 1000-2000, and Tenant B configures a trunk with the same provider but sets a DID pool of 2000-3000, this will create a conflict. Since both tenants are using the same trunk provider with overlapping DID pools, the PBX will not know how to route calls to DID number 2000.

DID Pool Numbering

The DID pool can include individual numbers or ranges of numbers, as illustrated below:

  • 1000-2000

  • 282556000-282556900

  • 101; 203; 300-450

DID numbers or ranges cannot begin with "+", "0", or "00" when adding them to the DID pool for a tenant. If your DID number or range starts with any of these characters, please remove them before entering.

Structuring the INVITE Message

The values included in outgoing INVITE messages sent by PortSIP PBX can be configured within the SIP trunk settings on the Outbound Parameters page. This allows you to customize the values assigned to each SIP field, which will be discussed in more detail later.

Below is an example of a basic INVITE message sent by PortSIP PBX when initiating an outgoing call through a SIP trunk:

INVITE sip:88888888@pstn.twillio.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-524287-1---39f4bc06b915b70f;rport
Max-Forwards: 19
Contact: <sip:101@192.168.0.11:5060;ob>
To: <88888888@pstn.twillio.com>
From: <sip:101@sip.portsip.io>;tag=35c2342b
Call-ID: DSSzmYKnCwhMeB1TWUwv2A..
CSeq: 1 INVITE
Session-Expires: 300
Min-SE: 90
Accept-Language: en
Allow: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE, PUBLISH
Content-Type: application/sdp
Supported: replaces, norefersub, tdialog, join, timer
User-Agent: PortSIP UC - Call Manager 16.0.0.302
X-CID: QERh8-brREchTYRqTSTALQ..
X-Session-Id: 678271070897180672
X-Trunk-Name: CallCentric
Content-Length: 361

Configurable SIP Fields

PortSIP PBX does not allow full customization of the INVITE message, meaning its base structure must always include the essential SIP fields shown above. However, PortSIP PBX allows the configuration of the following fields:

  • Request Line URI : User Part

  • Request Line URI : Host Part

  • Contact : User Part

  • Contact: Host Part

  • To : Display Name

  • To : User Part

  • To : Host Part

  • From : Display Name

  • From : User Part

  • From : Host Part

  • Remote Party ID - Called Party : Display Name

  • Remote Party ID - Called Party : User Part

  • Remote Party ID - Called Party : Host Part

  • Remote Party ID - Calling Party : Display Name

  • Remote Party ID - Calling Party : User Part

  • Remote Party ID - Calling Party : Host Part

  • P-Asserted Identity : Display Name

  • P-Asserted Identity : User Part

  • P-Asserted Identity : Host Part

  • P-Preferred Identity : Display Name

  • P-Preferred Identity : User Part

  • P-Preferred Identity : Host Part

  • P-Called-Party-ID : Display Name

  • P-Called-Party-ID : User Part

  • P-Called-Party-ID : Host Part

  • Privacy Types Supported

Variables

The variables briefly described below can be assigned to each of the SIP Fields mentioned in the section Configurable SIP Fields.

AuthID

The value of the AuthID variable is derived from the Authentication ID field located in Call Manager > Trunks > Edit Register Based Trunk > Client Verification tab > Authentication Name (SIP User ID).

This variable is only applicable for Register-Based Trunks and Accept Register-Based Trunks.

CalledNum

The CalledNum variable contains the dialed/destination number the INVITE message is supposed to initiate the call.

CallerNum

This variable refers to the user part of the SIP To header.

CallerDispName

When an extension user registered with PortSIP PBX makes a call to an external number, and a value is provided in the From: Display Name field, PortSIP PBX attempts to preserve this value in the INVITE message sent to the SIP trunk. If the endpoint does not use a display name, PortSIP PBX will default to using the extension's name for this variable.

OriginatorCallerID

The OriginatorCallerID variable attempts to retain the original caller's number, even if that number did not originate from PortSIP PBX. If an extension user registered with PortSIP PBX uses an anonymous dial code to make an outbound call, this variable will be populated with the value anonymous.

Anonymous

When the PBX sends the call to the trunk, this variable will be populated with the value anonymous.

Custom Field

Choosing this as a variable will allow you to enter a value manually.

TrunkHostPort

This variable is the combined value of the Host Domain or IP or IP and Port fields in the Trunks > Edit SIP Trunk > Server tab.

OutHostPort

This variable is the combined value of the Outbound Proxy Server and Outbound Proxy Server Port fields in the Trunks > Edit SIP Trunk > Server tab.

ContactUri

The "ContactUri" variable value containing the IP and port on which a SIP Trunk should contact PortSIP PBX, is typically used to populate the Contact : Host Part. The port is always the port of PortSIP transport.

OutboundCallerId

During an outbound call, the PBX checks for an Outbound Caller ID from various locations, with priority given to certain fields. The highest priority is the Outbound Caller ID field set in the extension settings.

To configure the Outbound Caller ID, navigate to Call Manager > Users > Edit User > Extension tab, and in the Outbound Caller ID section, you can set the outbound caller ID for the user.

The outbound caller ID must fall within the trunk's DID pool range, and the DID number can begin with "+", "0", "00", or "+00".

As explained in the User Groups section, you can create a user group, add users as group members, and assign DID numbers to the group. If an outbound rule is created using the Calls from user groups option and a group member does not have an outbound caller ID assigned, the PBX will automatically use the outbound caller ID from the user group when that member makes an outbound call.

If both the user’s outbound caller ID and the group’s outbound caller ID are empty, or if the outbound rule does not use user group criteria, the PBX will default to the company’s outbound caller ID settings. To configure the company-wide outbound caller ID, go to Company > Outbound Caller ID in the menu.

Outbound Caller ID Priority

When making an outbound call through a SIP trunk, the PBX applies the outbound caller ID in the following order of priority:

  1. App specified Outbound Caller ID: If the user makes a call from the PortSIP ONE app and specifies an Outbound caller ID, the PBX will use that caller ID anyway.

  2. Caller Extension's Outbound Caller ID: If the caller’s extension has an outbound caller ID set, the PBX will use this ID to overwrite the FROM header.

  3. Outbound Rule's Outbound Caller ID: If the caller extension’s outbound caller ID is not set, the PBX will use the outbound caller ID configured in the outbound rule to overwrite the FROM header.

  4. User Group's Outbound Caller ID: If the outbound rule does not have an outbound caller ID set but specifies a user group with an outbound caller ID, the PBX will use the group’s outbound caller ID to overwrite the FROM header.

  5. Company's Outbound Caller ID: If none of the above IDs are configured, the PBX will default to the company’s outbound caller ID to overwrite the FROM header.

Outbound Caller ID for PBX Services

You can configure the outbound caller ID for system services such as the Virtual Receptionist, Call Queue, Ring Group, and Meeting features in PortSIP PBX.

When calls from a call queue, ring group, or virtual receptionist are not answered, fail, or time out and are re-routed to the trunk, or when a meeting invites an external number to join via the trunk, the outbound caller ID will be taken from the respective service's outbound caller ID settings. If no specific outbound caller ID is configured for these services, the system will default to the company’s outbound caller ID.

Adjusting Trunk Parameters to Apply Outbound Caller ID

Refer to the screenshot below to adjust the trunk's outbound parameters and apply the outbound caller ID for outbound calls.

Bypassing the Outbound Caller ID Settings

PortSIP PBX allows you to bypass the default outbound caller ID settings by letting the caller’s device specify the outbound caller ID via a custom SIP header in the INVITE message. By adding the X-Outbound-Cli SIP header, the PBX can modify the INVITE message before sending it to the trunk.

Here’s how it works:

  • X-Outbound-Cli: rewrite-from=123 This header rewrites the username in the From header to 123 before the PBX sends the INVITE to the trunk.

  • X-Outbound-Cli: rewrite-pai=456 This header adds a P-Asserted-Identity SIP header, setting the username to 456 before forwarding the INVITE to the trunk.

  • X-Outbound-Cli: rewrite-rpi=789 This header adds a Remote-Party-ID SIP header, setting the username to 789 when the INVITE is sent to the trunk.

To apply these changes to a specific trunk, you can include the trunk ID in the header:

  • X-Outbound-Cli: rewrite-from=123; trunk-id=235468356 This ensures that the PBX rewrites the From header only when the call is routed through the trunk with the ID 235468356.

  • X-Outbound-Cli: rewrite-from=123; rewrite-rpi=789; rewrite-pai=456; trunk-id=235468356 With this header, the PBX will rewrite the From, Remote-Party-ID, and P-Asserted-Identity headers for the specified trunk.

If no trunk-id is specified, the PBX will apply these changes to any trunk that the call is routed through.

For the v16.x, you should use the trunk-name instead of the trunk-id.

When using the "X-Outbound-Cli" header to bypass the PBX outbound caller ID settings, the specified outbound caller ID must be in the trunk DID Pool range assigned to the tenant.

Remove the SIP Header

PortSIP PBX support allows you to remove the following SIP headers when sending the SIP message to the SIP trunk:

  • P-Asserted-Identity

  • P-Preferred-Identity

  • P-Called-Party-ID

Simply set up the SIP headers as shown in the following screenshot and the headers will be removed.

Privacy Types Supported

The PortSIP PBX supports the following Privacy types:

  • user: user-level privacy function provided. Any non-essential informational headers are removed, including the Subject, Call-Info, Organization, User-Agent, Reply-To, and In-Reply-To. Possibly the original value of the From header is changed to anonymous.

  • header: headers that cannot be set arbitrarily by the user (Contact/Via) are modified. No unnecessary headers that might reveal personal information about the originator of the request are added. (The values modified must be recoverable when further messages in the dialog need to be routed to the originator.)

  • id: third-party asserted identity kept private with respect to SIP entities outside the trust domain with which the user authenticated.

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