SIP Trunk Testing: How to Test a Trunk Before You Pay

Why Test Before Going Live?

Setting up a SIP trunk involves multiple moving parts: network configuration, firewall rules, codec negotiation, and dialplan routing. Testing with a $0-per-channel trunk lets you identify and fix issues without affecting your production phone service.

Why IPComms SIP Trunks Cost $0 Per Channel

Most SIP trunk providers charge $15–25 per channel per month before you even make a call. IPComms takes a different approach: the trunk itself carries no fees. You only pay for phone numbers and usage.

This is not a trial or a limited tier. It is our standard pricing model. There are no contracts, no trunk setup fees, and no per-channel charges. Stop worrying about SIP trunk capacity — there are no set call limits, so you can run concurrent channels at $0/month.

IPComms vs Typical SIP Trunk Providers

What You Get When You Test with IPComms

IPComms offers affordable pay-as-you-go pricing that makes it easy to test your PBX setup with minimal commitment:

Step 1: Sign Up for a Free Account

1. Go to portal.ipcomms.net
2. Click "Create Account"
3. Enter your email and create a password
4. Verify your email address
5. Select your plan and add a DID to get started

Once your account is created, you will see your SIP credentials and a test DID assigned to your account. Note down the following:

SIP Server:    s1.ipcomms.net
SIP Port:      5060 (UDP) / 5061 (TLS)
Server IP:     34.23.59.14
Your DID:      (shown in portal)
Username:      (shown in portal)
Password:      (shown in portal)

Step 2: Configure Your Trunk

Use the credentials from the portal to set up your PJSIP trunk. Here is a minimal configuration for testing:

; === Transport ===
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
; === IPComms Test Trunk ===
[ipcomms-test]
type=endpoint
transport=transport-udp
context=from-ipcomms
disallow=all
allow=ulaw
allow=alaw
outbound_auth=ipcomms-test-auth
aors=ipcomms-test
from_domain=s1.ipcomms.net
direct_media=no
[ipcomms-test]
type=aor
contact=sip:s1.ipcomms.net
qualify_frequency=60
[ipcomms-test-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD
[ipcomms-test-identify]
type=identify
endpoint=ipcomms-test
match=34.23.59.14
[ipcomms-test-reg]
type=registration
transport=transport-udp
outbound_auth=ipcomms-test-auth
server_uri=sip:s1.ipcomms.net
client_uri=sip:YOUR_USERNAME@s1.ipcomms.net
retry_interval=60

Basic Test Dialplan

[from-ipcomms]
; Inbound calls from IPComms test trunk
exten => _X.,1,NoOp(Test inbound: ${EXTEN} from ${CALLERID(num)})
 same => n,Answer()
 same => n,Playback(tt-weasels)
 same => n,Hangup()
[outbound-test]
; Outbound test calls
exten => _1NXXNXXXXXX,1,NoOp(Test outbound: ${EXTEN})
 same => n,Dial(PJSIP/${EXTEN}@ipcomms-test,60)
 same => n,Hangup()

Step 3: Make Test Calls

Once your trunk is configured and registered, run through these tests:

Test Inbound

1. Call your test DID from a cell phone
2. Verify Asterisk answers and plays the test audio
3. Check the Asterisk CLI for the call trace: asterisk -rx "core show channels"

Test Outbound

1. From an extension on your PBX, dial an external number (your cell phone)
2. Verify the call connects and you hear audio in both directions
3. Check caller ID shows your test DID on the receiving phone

Verify Registration

# Check registration status
asterisk -rx "pjsip show registrations"
# Should show:
# ipcomms-test-reg    s1.ipcomms.net    Registered
# Check endpoint status
asterisk -rx "pjsip show endpoint ipcomms-test"

Testing Checklist

Run through this checklist to ensure everything is working before going to production:

Inbound calls connect

Call your test DID and verify Asterisk receives it

Outbound calls connect

Dial an external number from your PBX

Two-way audio

No one-way audio issues (common NAT/firewall problem)

Caller ID correct

Outbound calls show your test DID as caller ID

DTMF works

Press digits during a call and verify they are detected

Call quality acceptable

No echo, choppy audio, or excessive delay

Concurrent calls work

Make 2 calls simultaneously to verify channel capacity

Failover (optional)

If using TLS, verify it negotiates correctly

Going Live After Testing

Once all tests pass, you are ready to go live. Here is the upgrade path:

1. Upgrade your account in the portal to a paid plan
2. Order your production DIDs or port your existing numbers
3. Update your trunk config - credentials stay the same, just update your dialplan for the new DIDs
4. Switch to IP authentication if you have a static IP (recommended for production)
5. Enable TLS/SRTP for encrypted calls