Trunks

Open

You can edit, clone or delete a trunk clicking the three dots button (⋮) next to the trunk of interest. You can access the realtime panel to monitor trunk activity.

Furthermore, under Tools → Trunks, you can:

• search for a specific trunk

• clear all filters

• manage columns, by selecting or not them

• activate the advanced search for each field

• create a new trunk

2. Enter the Trunk’s details:

• Name

• Domain/Host name (please refer to your provider)

• Authentication Password

• Default User Name

• Trunk Registry string (which is usually defaultuser:password@host)

3. Click the Active flag to enable it

4. Click Add

• Active: enable or disable the trunk

• Host: domain or host name

• Secret: Authentication password

• Default User: Authentication user

• Context: the applicable context

• Caller ID: The Caller ID in the format "name"

• Type:

  • "user" is matched on username (in the From: header)

  • "peer" is matched on IP address

  • "friend" allows both, but can introduce matching ambiguity

• DTMF Mode: how DTMF (Dual-Tone Multi-Frequency) tones are transmitted:

  • RFC2833 (default): DTMF tones are sent via RTP, outside the audio stream

  • INBAND: DTMF is sent within the audio stream and is audible. This method requires high CPU usage

  • INFO: DTMF is sent using SIP INFO messages. Reliable but not widely supported by all PBX systems and SIP trunks

• NAT: adjusts Asterisk behavior for clients behind firewalls.
  If any of the comma-separated options is set to no, Asterisk will override other settings and set nat=no.

• Qualify: determines when the SIP is achievable

• Allowed Codecs: specifies the list of supported codecs, ordered by priority

• Insecure: defines how Asterisk handles connections from peers with insecure settings. Useful for compatibility with certain SIP trunks.

• Calls Limit: maximum number of concurrent inbound and outbound calls allowed

• Optional Description

• Registry: required for receiving inbound calls when authentication is needed

• Direct Media: Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT).

  • The default setting is YES.

  • If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off.

• Call Counter: enables call counters on devices

• From Domain: sets the default From:domain in SIP messages when Asterisk acts as a SIP user agent. Some providers require this to be a valid domain instead of just an IP address

• From User: defines the username to be sent in the From header when placing calls to the peer (another SIP proxy). Used for authentication. Valid only for type=peer.

3. Check the information in realtime:

• Trunk Status, which shows status of a SIP trunk registration to motion server

• Registry, which shows SIP registration status when Motion registers as a client to the provider.