GoAutoDial SIP Trunk Setup: Carrier Entry Basics

GoAutoDial looks friendlier than a bare Vicidial install, and it is. Under the hood, though, it's still Vicidial and Asterisk doing the heavy lifting. So when a new GoAutoDial box "won't dial," the cause is almost always the same handful of fields on the carrier screen: the account entry, the registration string, the dialplan entry and the globals string.

This guide walks through a GoAutoDial SIP trunk setup: what to get from your carrier, where the carrier screen lives in v4 and v3, what goes in each field, and how to test. Every IP, hostname and credential below is a placeholder.

Example values used in this guide (placeholders only)

• Carrier signaling IP: 203.0.113.10
• Carrier hostname: sip.carrier.example.com
• Your dialer's public IP: 203.0.113.50
• Carrier ID / peer name: KINGTRUNK

GoAutoDial in One Minute: Vicidial with a Different Front Door

GoAutoDial is an open-source contact center suite built on Vicidial and Asterisk. It adds its own admin interface, but carriers still use Vicidial's carrier model. So if you've ever added a SIP trunk carrier in Vicidial, most of what you know carries straight over.

The two versions you'll run into most:

• GoAutoDial v4 was released as a CentOS 7 based installer with Asterisk 13, Vicidial 2.14, Kamailio and WebRTC browser phones. Carriers live in the GoAutoDial admin under Settings → Carriers.
• GoAutoDial v3 (CE 3.x) shipped with its own admin plus the classic Vicidial admin. You can add carriers in the classic screen at Admin → Carriers → Add A New Carrier, and the v3 admin has its own carrier screens too. Menu names vary between builds, so if something isn't where we say, look for "Carriers."

Worth knowing: CentOS 7 reached end of life in June 2024. If your GoAutoDial box still runs on it, plan for security patching alongside your carrier work.

What to Get from Your Carrier First

Before you touch the admin, ask your SIP trunk provider for:

• Authentication method: IP authentication or registration (username and password).
• Signaling addresses: the IP or hostname and port to send calls to, plus every IP they may send inbound calls from.
• Codecs: usually G.711 µ-law (ulaw) for US traffic, sometimes G.729.
• DTMF method: usually RFC 2833.
• Number format: 1+10 digits, 10 digits, E.164 with +, or a tech prefix.
• Caller ID rules: the format they expect and which numbers you're allowed to present.
• Media (RTP) IP ranges, so you can open your firewall correctly.

IP Authentication or Registration?

With a fixed IP, IP authentication is usually the sturdier choice: nothing to re-register, no password to leak.

Step-by-Step GoAutoDial Carrier Setup

Step 1: Open the Carrier Wizard (v4)

In GoAutoDial v4, go to Settings → Carriers and click the + button. The Carrier Wizard asks you to choose a carrier type. Pick Manual for a third-party SIP trunk. (The other choices cover copying an existing carrier and a bundled provider option.)

Then fill in the basics:

• Carrier ID: short, no spaces, for example KINGTRUNK.
• Carrier Name, Description, User Group: labels and who can use it.
• Authentication: IP Based or Registration. Registration asks for username, password, server IP/host and port.
• SIP Server, Dial Prefix, Codecs, DTMF Mode: from your carrier's details. The dial prefix is the digit (often 9) your dialplan expects in front of each number.
• Protocol: SIP for a normal SIP trunk.
• Server IP: the GoAutoDial server that should build this carrier.
• Active: Yes when you're ready to push it live.

The wizard writes the underlying Vicidial fields for you. Afterward, open the carrier's advanced configuration view and review the account entry, dialplan entry and globals string. Wizards are helpful, not psychic.

On v3, or if you prefer the raw form, use the classic Vicidial screen (Admin → Carriers → Add A New Carrier) and fill in the same fields by hand.

Step 2: The Account Entry

The account entry is the SIP peer Asterisk uses to reach your carrier. GoAutoDial v4 ships Asterisk 13 and builds carriers as classic chan_sip peers.

Example: IP authentication (placeholder values)

[KINGTRUNK]
type=peer
host=203.0.113.10
port=5060
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=trunkinbound
insecure=port,invite
qualify=yes
nat=force_rport,comedia

Example: registration (placeholder values)

[KINGTRUNK]
type=friend
host=sip.carrier.example.com
username=YOUR_USERNAME
secret=YOUR_PASSWORD
fromuser=YOUR_USERNAME
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=trunkinbound
insecure=port,invite
qualify=yes
nat=force_rport,comedia

Keep context=trunkinbound so inbound calls land where your DIDs are matched to in-groups. Older community samples use insecure=very and nat=yes. Those options are deprecated in newer Asterisk releases, so stick with the explicit values above.

Step 3: The Registration String (Registration Only)

If your carrier uses IP authentication, leave this field blank. If it uses registration, the line follows the standard chan_sip format:

register => YOUR_USERNAME:YOUR_PASSWORD@sip.carrier.example.com:5060

Some carriers want the username repeated after a slash (.../YOUR_USERNAME). Follow your carrier's sample.

Step 4: The Globals String

The globals string maps a short variable to your trunk so the dialplan stays tidy:

KINGTRUNK=SIP/KINGTRUNK

The name after SIP/ must match the bracketed name in your account entry. It's optional, but it makes swapping carriers later easier.

Step 5: The Dialplan Entry

This is where dialed numbers meet the trunk. A standard US entry, using dial prefix 9:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${KINGTRUNK}/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,n,Hangup()

What each piece does:

• _91NXXNXXXXXX matches the 9 dial prefix, the 1 country code and a 10-digit North American number.
• AGI(...call_log) logs the call for Vicidial's reports. Keep it as the first step.
• ${EXTEN:1} strips the 9, so the carrier receives 1NXXNXXXXXX. Need 10 digits? Use ${EXTEN:2}. Need a tech prefix? Put it in front.
• tTo are Dial options. Keep the o, which Vicidial relies on to pass the original caller ID.

Avoid catch-all patterns like _X. on production; they'll send anything, typos included.

Step 6: Point Your Campaign at the Carrier

In v4, go to Telephony → Campaigns, open your campaign, and set Carrier to use for this Campaign to your new carrier (only active carriers appear in the list). In the classic Vicidial admin, set the campaign's Dial Prefix to match your dialplan.

Then set a Campaign CallerID you're authorized to use, and let the campaign and list settings manage caller ID rather than the carrier dialplan.

About caller ID and STIR/SHAKEN: your carrier signs calls under STIR/SHAKEN, which helps authenticate that you're allowed to use the caller ID you send. It doesn't decide labeling. Our STIR/SHAKEN guide explains how attestation and labels relate.

Step 7: Test Before Agents Log In

From the Asterisk CLI (asterisk -rvvv):

• sip show peers: look for KINGTRUNK with an OK status.
• sip show registry: for registration trunks, look for Registered.
• Place one manual test call and watch the CLI. A 403 usually means an IP or credential mismatch. A 404 or 484 usually means number format. One-way or no audio usually means firewall, NAT or RTP ports.

After go-live, keep watching ASR and ACD on the new carrier. Our guide to ASR and ACD for call centers explains how to read both numbers.

A v4-specific gotcha: on the stock v4 build, Kamailio listens on UDP 5060 for the WebRTC agent phones, and Asterisk listens on UDP 5070. Outbound calls to your carrier still go to the carrier's normal port, but inbound calls must reach Asterisk. If inbound calls ring nowhere, check which port your carrier is sending to and what your firewall allows.

Common GoAutoDial Carrier Mistakes

1. Peer name mismatch. The bracketed name, the globals string and the Dial() line must agree.
2. Wrong digit stripping. ${EXTEN:1} vs ${EXTEN:2} is the difference between connecting and a 404.
3. Carrier left inactive. An inactive carrier won't appear in the campaign carrier list.
4. Firewall gaps. Allow the carrier's signaling and media ranges, and nothing broader than you need.

Heading toward a newer Vicidial build later? Asterisk removed chan_sip in Asterisk 21, so carriers eventually move to PJSIP. Our chan_sip to PJSIP carrier migration guide covers running both stacks, translating the peer and cutting over safely.

A Quick Compliance Note

A working trunk isn't a license to dial. Outbound calling must follow the TCPA, the Telemarketing Sales Rule, the National Do Not Call Registry and state rules. Get proper consent for the type of call and technology you use, scrub your lists and honor opt-outs. This isn't legal advice; talk to your counsel.

Where Connectivity King Fits

Our SIP trunks for call centers work with GoAutoDial, Vicidial, Asterisk/FreePBX and 3CX. You get Tier-1 routes on a network built for 99.99% uptime, plus US DID numbers for caller ID and inbound. We'll send you the exact account entry and dialplan values for your setup, and our support team can help you test, Monday to Friday, 9am ET to 9pm PT.

Rather not babysit servers? A hosted Vicidial dialer comes with VoIP and DIDs connected. Deciding? Read hosted Vicidial vs self-hosting.

GoAutoDial SIP Trunk FAQ

Get Your GoAutoDial Trunk Crowned

Ready for carrier-grade SIP trunks that drop straight into GoAutoDial? We offer Tier-1 routes and DIDs across the United States, and call centers based anywhere can sign up. Long live the call that connects. Request a quote. Quotes usually come back the same day. You can also reach us at sales@connectivityking.com or +1 (888) 885-4202, Monday to Friday, 9am ET to 9pm PT.

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