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 free trunk lets you identify and fix issues without affecting your production phone service.
Verify Connectivity
Confirm SIP traffic passes through your firewall and NAT correctly.
Test Audio Quality
Check for one-way audio, echo, jitter, or codec issues before going live.
Validate Dialplan
Make sure inbound and outbound call routing works as expected.
Zero Risk
No impact on your existing phone service. Test in parallel with your current provider.
What You Get with IPComms Free Testing
IPComms offers affordable pay-as-you-go pricing that makes it easy to test your PBX setup with minimal commitment:
| Feature | Trial Includes |
|---|---|
| Test DID (phone number) | 1 local number in your area code |
| Outbound calling | US/Canada calls included |
| Concurrent channels | 2 simultaneous calls |
| Codecs | G.711 (ulaw/alaw), G.729 |
| TLS/SRTP | Full encryption available |
| Authentication | IP auth or registration |
| Credit card required | No |
No Strings Attached: The trial does not auto-convert to a paid account. When you are ready to go live, you upgrade manually and port your numbers.
Step 1: Sign Up for a Free Account
- Go to portal.ipcomms.net
- Click "Create Account"
- Enter your email and create a password
- Verify your email address
- 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=60Basic 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()Quick Start: For a more detailed trunk setup with TLS, SRTP, and IP authentication, see our full PJSIP trunk setup guide.
Step 3: Make Test Calls
Once your trunk is configured and registered, run through these tests:
Test Inbound
- Call your test DID from a cell phone
- Verify Asterisk answers and plays the test audio
- Check the Asterisk CLI for the call trace:
asterisk -rx "core show channels"
Test Outbound
- From an extension on your PBX, dial an external number (your cell phone)
- Verify the call connects and you hear audio in both directions
- 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:
- Upgrade your account in the portal to a paid plan
- Order your production DIDs or port your existing numbers
- Update your trunk config - credentials stay the same, just update your dialplan for the new DIDs
- Switch to IP authentication if you have a static IP (recommended for production)
- Enable TLS/SRTP for encrypted calls
Seamless Transition: Your trunk credentials and configuration stay the same when upgrading. Just update the DID in your dialplan and you are live.
Get Started with IPComms Today
Get a free test trunk with a DID in your area code. No credit card required. Test your PBX configuration with real calls.