SIP Trunk Settings

Use this page as a reference for the fields you’ll encounter when you create or edit a SIP (Session Initiation Protocol) trunk, the connection that links your phone system to your telephony provider. The settings are grouped the same way they appear in the trunk editor: Basic, Advanced, DIDs/DDIs, Inbound Caller ID Reformatting, Outbound Caller ID, and SIP Headers.

Basic settings

Basic

Setting	What it does
Name	A label that helps you recognize the trunk in lists and routes.
Trunk Status	Turns the trunk on or off.
Select ITSP Template	The country of your internet telephony service provider (ITSP). If no template exists for your provider, choose General.
ITSP	Your provider, chosen from the list of certified SIP trunk providers.

SIP registration timer

Note

These fields apply only to shared register trunks. Shared trunks are configured and managed centrally, so the values here are read-only and cannot be changed in the phone system portal.

Setting	What it does
Registration Attempts	How many times the system retries after a registration failure. Enter 0 for unlimited retries.
Default Registration Time(s)	The registration validity period, in seconds. The system re-registers before this period ends so the connection stays up.

Detailed configuration

Trunk Type: choose the kind of trunk you are building:

• Register Trunk
• Peer Trunk (DID Based)
• Peer Trunk (Port Based), Cloud Voice supports up to 5 port-based peer trunks.
• Peer Trunk (Private Network)
• Account Trunk

The remaining connection fields depend on the trunk type you select.

Register Trunk

Setting	What it does
Transport	The transport protocol your ITSP provides. To use TCP, first turn on the SIP TCP port under PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Hostname/IP	Your ITSP’s IP address or domain.
Port	The SIP port your ITSP provides.
Domain	The domain used in the SIP URI of headers such as From and To. If your ITSP doesn’t provide one, enter the same value as Hostname/IP.
Username	The username used to register with the ITSP.
Password	The password that goes with the username.
Authentication Name	The authentication name used to register with the ITSP.
Enable Outbound Proxy	When on, all SIP packets for outbound calls placed over this trunk are sent to an outbound proxy server. Confirm your ITSP supports an outbound proxy and follow their guidance before configuring it.

Peer Trunk (DID Based / Port Based / Private Network)

The static IP address and its port are assigned by the system and cannot be edited:

Setting	What it does
Static IP Address	Assigned automatically. Read-only.
Port	Assigned automatically. Read-only.

Enter your provider’s details in the remaining fields:

Setting	What it does
Transport	The transport protocol your ITSP provides. To use TCP, first turn on the SIP TCP port under PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Hostname/IP	Your ITSP’s IP address or domain.
Port	The SIP port your ITSP provides.
Domain	The domain used in the SIP URI of headers such as From and To. If your ITSP doesn’t provide one, enter the same value as Hostname/IP.

Account Trunk

Setting	What it does
Transport	The transport a third-party device uses to register. To use TCP, first turn on the SIP TCP port under PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Username	A username for the trunk. This value doubles as the trunk number.
Password	A password that goes with the username.
Authentication Name	An authentication name for the third-party device to register with.

Advanced settings

Caution

The advanced settings assume a working knowledge of the SIP protocol, and incorrect values can disrupt calling. In most cases the defaults are correct; change only the fields your situation requires.

The Advanced tab covers codec selection, VoIP behavior, call restriction, and STIR/SHAKEN.

Codec

Every new trunk starts with a default preferred-codec list, which may not match what your ITSP supports. Adjust the list to match your provider’s codecs so you get the best call quality for the bandwidth used. Cloud Voice supports these codecs:

u-law, a-law, G729A, GSM, H264, H261, H263, H263P, iLBC, G722, G726, SPEEX, ADPCM, MPEG4, VP8, and Opus.

VoIP

DTMF Mode: how touch-tone (Dual-Tone Multi-Frequency, or DTMF) digits are sent:

• RFC4733 (RFC2833): digits travel in the RTP (Real-time Transport Protocol) stream as separate RTP packets rather than in the audio.
• Info: digits are carried in SIP INFO messages.
• Inband: digits are encoded into the audio signal.
• Auto: the system uses RFC4733 (RFC2833) if the device supports it, and falls back to Inband if not.

Qualify: periodically sends a SIP OPTIONS packet to check that the device is reachable.

Enable SRTP: turns encrypted RTP on or off for the trunk.

T.38 Support: enables or disables T.38 fax on the trunk. T.38 adds processing overhead, so leave it disabled unless you need it.

Inband Progress: controls how ringing is signaled to extensions that call out over this trunk:

• Selected, the system sends a 183 Session Progress when told to indicate ringing and immediately sends ringing as audio.
• Cleared, the system sends a 180 Ringing when told to indicate ringing and does not send it as audio.

To set inband progress globally, contact support to have a custom configuration file applied.

Ignore 183 Message without SDP: controls how the system handles 183 messages that carry no SDP (Session Description Protocol, the part of a SIP message that describes the call’s media):

• Selected, the system does not forward 183 messages without SDP.
• Cleared, the system converts 183 messages without SDP into ones with SDP and forwards them.

Forward the 180 (SDP) Message Following the Peer’s Format: controls whether a 180 message with SDP is forwarded, based on the 180 received from the other party:

• Selected, the system forwards a 180 with SDP when the received 180 includes SDP. This has no effect when Inband Progress is enabled.
• Cleared, the system does not forward a 180 with SDP, even when the received 180 contains it.

Dedicated Trunk: turn this on if your provider requires a dedicated internal IP address, then contact them to set up the related network configuration. Available only for register trunks, Peer Trunk (DID Based), and Peer Trunk (Port Based).

Enable SIP Authentication Cache: caches successful SIP authentication credentials and reuses them for later INVITE and BYE requests within the same session or registration cycle. Available only for register trunks.

Force SIP URI Scheme: forces the sip: URI scheme instead of sips: in registration requests, switching the SIP transport from encrypted to unencrypted. Available only for register trunks that use TLS or DNS-NAPTR transport.

Caution

Turning this on downgrades registration from encrypted to unencrypted. Enable it only if your provider requires the sip: scheme, and be aware that credentials and signaling will no longer be protected by TLS.

Authentication Error SIP Code: the SIP error code your ITSP returns to signal an authentication error. Available only for register trunks. Enter up to 10 codes, separated by ;.

Authentication Registration Attempts: how many times the system tries to register after receiving the authentication error SIP code from the ITSP. Available only for register trunks. The count includes the first attempt made without authentication information.

Global Registration Retry Interval (s): the interval, in seconds, before retrying registration after any non-200 SIP error code from the ITSP. Available only for register trunks.

Ignore 100 Trying Response: controls whether the system ignores the 100 Trying response. Available only for register trunks and account trunks.

Support SIP REFER: controls whether the system acts on SIP REFER requests received over the trunk. When enabled, use the Allow transfer to: list to permit Internal Number, External Number, or both as transfer destinations, and the system transfers the call as the REFER request instructs. For external numbers, the system prepends the configured prefix to the target number in the REFER request. Only blind transfers are supported.

Outbound Failover SIP Code: the SIP response codes that trigger failover to the next trunk in the same outbound route and re-initiate the call. If these codes conflict with other SIP settings, the codes here take precedence.

• Default: uses the built-in codes: 408, 500, 502, 503, 513, 555, and 580.
• Custom: lets you define your own codes. Only codes in the 300-699 range take effect, up to 32 codes, separated by ;.

Call restriction

Setting	What it does
Call Restriction Type	Which calls count toward the concurrent-call limit. Outbound Call restricts outbound calls only; All restricts both outbound and inbound calls.
Maximum Concurrent Calls	The most concurrent calls the trunk allows. The default is Unlimited.

Caution

If you set Call Restriction Type to All with a low Maximum Concurrent Calls value, incoming calls are rejected once the limit is reached. Size the limit to your trunk’s real capacity, and use Outbound Call if you only mean to cap outbound calling.

STIR/SHAKEN

STIR/SHAKEN is an industry-standard framework that helps prevent caller ID spoofing. For shared trunks it is configured centrally and applied to the associated systems, so you can view these settings in the portal but cannot change them.

STIR/SHAKEN Mode: the operating mode for calls over the trunk:

• Outbound Signing: the system digitally signs every outbound call over the trunk, except emergency and anonymous calls.
• Inbound Verification: the system verifies every inbound call over the trunk and rejects calls according to the rejection criteria configured globally for shared trunks.
• Signing & Verification: the system both signs outbound calls and verifies inbound calls.

Upstream Verification Result Handling: when enabled, the system trusts the ITSP’s verification result and skips its own independent verification.

Verification Status Parameter in PAI Header: the parameter name within the P-Asserted-Identity (PAI) header where the ITSP places the signature verification result for inbound calls.

Header Field for SHAKEN Attestation Level: the parameter name the ITSP uses to convey the SHAKEN attestation level of inbound calls.

Enable Call Filtering: when enabled, the system uses the ITSP’s verification results and rejects calls according to the rejection criteria configured on the shared trunk.

Drop Calls by Verification Status: the verification statuses that cause a call to be rejected.

DIDs/DDIs

Direct Inward Dialling (DID), also called Direct Dial-in (DDI), is a service offered by telephone companies. For background, see DID Number Overview.

• DID numbers are commonly used on inbound routes to distinguish incoming calls. See Route Inbound Calls based on DID Numbers.
• For steps to set up DID numbers on this trunk, see Configure DID Numbers on a Trunk.

Inbound caller ID reformatting

Sometimes the caller ID your provider sends isn’t in a format you can redial directly. You can reformat the inbound caller ID on a per-trunk basis; the change is applied before the caller ID reaches the called party. For details and examples, see Reformat Inbound Caller ID based on a Trunk.

Outbound caller ID

The outbound caller ID is the number or name shown on the called party’s device. You can set one global outbound caller ID for the trunk or assign caller IDs to individual extension users. When a caller ID number is set, callers see that number instead of the extension’s own number.

Note

Every trunk has a default number that appears on the called party’s device. Custom outbound caller ID requires support from your trunk provider, so contact them before configuring it. Otherwise the setting won’t take effect and outbound calls may fail.

For the different ways to configure outbound caller ID, see Customize Outbound Caller IDs for Outbound Calls.

SIP headers

Caution

The SIP header settings assume a working knowledge of the SIP protocol, and incorrect values can disrupt calling. Keep the defaults unless a specific field needs to change.

The SIP Headers tab covers inbound parameters, outbound parameters, custom SIP headers, and other settings.

Inbound parameters

Get Caller ID From: the header field the trunk reads the caller ID from:

• Follow System: use the global Get Caller ID From setting.
• From
• Contact
• Remote-Party-ID
• P-Asserted Identify
• P-Preferred-Identity

Get DID From: the header field the trunk reads the DID number from. Because providers place the DID in different headers, an incoming call fails if the system reads the wrong one. Inspect the SIP packets from your provider before choosing:

• Follow System: use the global Get DID From setting.
• To
• Invite
• Diversion
• Remote-Party-ID, if you choose this but the provider doesn’t support Remote Party ID, the system reads the DID from the INVITE header instead.
• P-Asserted Identify
• P-Called-Party-ID
• P-Preferred-Identity

Outbound parameters

For outbound calls, you can define the parameters carried in the SIP INVITE headers. The available values for many of these are listed in Options for outbound parameters below. Which optional parameters appear depends on the trunk type.

From User Part: the caller ID placed in the SIP From header.

From Display Name Part: the caller ID name placed in the SIP From header.

From Host Part: the domain or IP address used in the From field of the SIP INVITE header. Available only for peer trunks; set it to match your provider’s requirements, or calls may fail.

• Default: use the domain or IP address from the Domain field of the peer trunk.
• Custom: enter your own domain or IP address in the field beside the dropdown.

To Host Part: the domain or IP address used in the To field of the SIP INVITE header. Available only for peer trunks; set it to match your provider’s requirements, or calls may fail.

• Default: use the domain or IP address from the Domain field of the peer trunk.
• Custom: enter your own domain or IP address in the field beside the dropdown.

Diversion, Remote-Party-ID, P-Asserted-Identity, P-Preferred-Identity: optional INVITE-header parameters you can set as needed.

P-Asserted-Identity URI Format: the format used to assert and verify the caller’s identity: SIP URI (sip:) or TEL URI (tel:). Available only when P-Asserted-Identity is not set to None.

Options for outbound parameters

Option	What it uses
[Default]	For Diversion, the system uses the original call destination number (the DID number) when call diversion occurs. For Remote-Party-ID, P-Asserted-Identity, and P-Preferred-Identity, the system picks a value using this priority, top to bottom: Outbound Route Outbound Caller ID, Extension’s Outbound Caller ID in Trunk, Trunk Outbound Caller ID, Trunk Username, Extension Caller ID, and finally the Originator Caller ID.
[None]	The parameter is left out of the SIP INVITE packet.
Extension Caller ID	The caller ID set on the extension.
Trunk Outbound Caller ID	The trunk’s global outbound caller ID (Trunk > Outbound Caller ID > General).
Extension’s Outbound Caller ID in Trunk	The extension’s outbound caller ID associated with this trunk. Direct outward dialing (DOD) is a number an extension can present as its caller ID on outbound calls. If the extension selects a specific DOD, that DOD number is used. If it uses the default DOD, the associated DOD number is used; when several DOD numbers are associated, the highest-priority one in the extension’s DOD list wins (Extension and Trunk > Extension > User > Outbound Caller ID (DOD) > Outbound Caller IDs).
Outbound Route Outbound Caller ID	The outbound caller ID set on the outbound route used for the call.
Originator Caller ID	The caller ID of the call originator, the first caller when a call is transferred. If the originator is an external number, that number is used; if it’s an extension, the order is Extension Outbound Caller ID, then [Default].
Trunk Username	The username set on the trunk.
Custom	A value you define yourself.

Custom SIP headers

Note

Custom SIP header settings apply only to shared trunks. Shared trunks are managed centrally, so these values cannot be changed in the phone system portal.

Setting	What it does
Custom SIP Header	The name of the custom SIP header. The system includes it in SIP INVITE messages for outbound calls over this trunk.
Value	The value of the custom SIP header.

Other settings

User Agent: if your ITSP requires a User Agent for authentication, enter the value they provide.

Realm: a string shown to users so they know which username and password to use. If you’re unsure what to enter, ask your provider.

100rel: how the trunk handles reliable provisional responses: Required, Supported, or Disabled.

Maxptime: the maxptime value the system sends in the INVITE packet. With [Default], the system sends a maxptime that matches the codec used for the outbound call.

Send Privacy ID: whether to include the Privacy ID in the SIP header. Off by default.

User Phone: whether to add user=phone to the request line in the SIP INVITE header. Enable only if your provider requires it.

Send X-OpenAPI-Call-ID: whether to include an X-OpenAPI-Call-ID field in the SIP INVITE packet to carry the Call ID for inbound and outbound calls over the trunk.

Support P-Early-Media: whether the P-Early-Media field is included in the INVITE packet.

Send 183 Message with P-Early-Media Header: whether the system includes a P-Early-Media header with the value sendrecv in the 183 message for inbound calls over the trunk.

Force Selected DOD in From Header: whether the system ignores the trunk’s From User Part setting and instead uses the extension’s selected DOD number in the From header when the extension selects a specific DOD for an outbound call.

Return 302 on External Forward: whether the system returns a “302 Moved Temporarily” response to the caller when forwarding an inbound call to an external number, letting the carrier redirect the call directly.