SIP Trunk Setup: VICIdial, Asterisk & FreePBX

Configuring a SIP trunk is the same fundamental process on every platform: point your phone system at the carrier’s SIP proxy, authenticate, set your codec and DTMF preferences, handle NAT if you are behind a firewall, and test. The specifics differ by platform — Asterisk uses sip.conf text files, FreePBX uses a web GUI, and VICIdial adds a carrier management layer on top of Asterisk. This guide walks through all three.

For detailed VICIdial-specific configuration including campaign setup, CID groups, and dial plan, see our comprehensive VICIdial SIP trunk setup guide. This guide covers the trunk connection itself across all three platforms.

SIPNEX provides SIP trunking for all three platforms. The carrier-side configuration is identical — the differences are entirely in how each platform manages its SIP settings.

What you need from your carrier

Before configuring any platform, you need these details from your SIP trunk provider:

• SIP Proxy address: The IP address or hostname of the carrier’s SIP server
• Authentication method: IP-based (carrier whitelists your IP) or digest (username/password)
• If digest: username and password
• Supported codecs: Typically G.711u, G.711a, G.729
• DTMF method: Typically RFC 2833
• Port: Usually 5060 (SIP) or 5061 (TLS/SIPS)
• RTP port range: Usually 10000-20000 — needed for firewall rules

SIPNEX provides all of this during provisioning. We support both IP-based and digest authentication, G.711u as primary codec, and RFC 2833 for DTMF.

Asterisk (sip.conf)

Asterisk is configured through text files. The SIP trunk is defined as a peer in /etc/asterisk/sip.conf.

IP-based authentication:

[sipnex]
type=peer
host=CARRIER_IP
port=5060
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
canreinvite=no
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
insecure=invite
context=from-trunk

Digest authentication — add credentials and a register line:

[sipnex]
type=peer
host=CARRIER_IP
port=5060
username=YOUR_USER
secret=YOUR_PASS
fromuser=YOUR_USER
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
canreinvite=no
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
insecure=invite,port
context=from-trunk

register => YOUR_USER:YOUR_PASS@CARRIER_IP/sipnex

Outbound dial plan in /etc/asterisk/extensions.conf:

[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sipnex,,tTo)
exten => _1NXXNXXXXXX,2,Hangup()

After editing, reload Asterisk: asterisk -rx "sip reload" then verify: asterisk -rx "sip show peers" — the sipnex peer should show as OK or Reachable.

FreePBX (web GUI)

FreePBX provides a graphical interface over Asterisk. The trunk is configured through the web admin panel.

1. Navigate to Connectivity → Trunks → Add Trunk → Add SIP Trunk
2. Trunk Name: SIPNEX
3. Outbound CallerID: Your primary DID in E.164 format
4. Peer Details (under SIP Settings → Outgoing):

host=CARRIER_IP
port=5060
type=peer
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
canreinvite=no
nat=force_rport,comedia
qualify=yes
insecure=invite

5. For digest auth, add username=YOUR_USER and secret=YOUR_PASS to peer details, and add the register string in the Registration field: YOUR_USER:YOUR_PASS@CARRIER_IP

6. Submit and Apply Config

7. Configure an Outbound Route (Connectivity → Outbound Routes) that matches your dial pattern (11-digit US numbers: 1NXXNXXXXXX) and routes through the SIPNEX trunk.

8. Configure Inbound Routes (Connectivity → Inbound Routes) for each DID to route to the appropriate destination (extension, ring group, IVR).

VICIdial

VICIdial adds its own carrier management layer on top of Asterisk. The trunk is configured in both sip.conf (Asterisk level) and the VICIdial admin panel.

1. Add the sip.conf peer definition (same as Asterisk above)
2. In VICIdial Admin → Carriers → Add A New Carrier:
   • Enter the sip.conf peer block in the Account Entry field
   • Enter the outbound dial plan in the Dialplan Entry field
   • Set registration strings if using digest auth
3. Configure Campaigns to use the carrier for outbound dialing
4. Configure Inbound DIDs to route incoming calls to in-groups

See our complete VICIdial SIP trunk setup guide for detailed instructions including campaign configuration, CID groups for local presence, AMD settings, and troubleshooting.

Firewall configuration (all platforms)

Regardless of platform, your firewall must allow:

• Outbound UDP 5060 to your carrier’s SIP proxy IP (SIP signaling)
• Inbound UDP 5060 from your carrier’s SIP proxy IP (SIP signaling)
• Outbound/Inbound UDP 10000-20000 to/from your carrier’s media IPs (RTP audio)
• If using TLS: TCP 5061 to/from carrier

Block all other SIP traffic. An open SIP port on the public internet attracts brute-force registration attacks within hours. Only allow SIP traffic from known carrier IPs.

Testing your trunk

After configuration, test these scenarios:

1. Outbound call to a known number: Verify the call connects, audio is clear in both directions, DTMF works (press a key during the call and verify it is received)
2. Inbound call to your DID: Call your DID from an external phone. Verify it rings the correct destination, audio is bidirectional, caller ID displays correctly
3. Caller ID display: Place an outbound call and check what CID the recipient sees. Verify it matches the DID you configured
4. SIP registration (digest auth): Check sip show registry in Asterisk CLI. Status should show “Registered”
5. Trunk status: Check sip show peers in Asterisk CLI. Your carrier peer should show “OK” with qualify response time

If any test fails, check your Asterisk SIP debug logs (sip set debug on in the CLI) and look for the SIP response codes that indicate the failure type.

Frequently asked questions

SIPNEX provides trunk credentials and configuration support for Asterisk, FreePBX, VICIdial, and any SIP-compliant platform. Our support team has configured these platforms on their own systems. Get trunk credentials or see our rates.