“No password? No problem. Let your IP do the talking.”
If you’ve added your server’s public IP in the Bitcall panel, you can set up a direct SIP trunk without credentials.
🧰 What You’ll Need
A working Asterisk server
Your public IP address added to Bitcall’s Trusted IP list
⚠️ Your IP must be static and public (not behind NAT) for this to work properly.
🔧 Step-by-Step (chan_sip)
1. SSH into your server
ssh root@your-server-ip
2. Open
sip.conf
nano /etc/asterisk/sip.conf
Add the following at the bottom:
[bitcall-ip]
type=peer
host=gateway.bitcall.io
port=5060
context=outbound-bitcall
insecure=invite
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
nat=no
qualify=yes
📸 [Insert screenshot: sip.conf IP-auth peer config]
3. Open
extensions.conf
nano /etc/asterisk/extensions.conf
Add this to define your outbound dial context:
[outbound-bitcall]
exten => _X.,1,Set(CALLERID(num)=your-callerid)
same => n,Dial(SIP/${EXTEN}@bitcall-ip,60)
same => n,Hangup()
⚠️ Replace your-callerid with a valid CLI registered in Bitcall.
📸 [Insert screenshot: extensions.conf with outbound context]
4. Reload Asterisk
asterisk -rx "reload"
And check your console:
asterisk -rvvv
You won’t see “Registered” (since IP-auth doesn’t use REGISTER), but you’re now ready to make outbound calls.
🧪 Testing a Call
From a SIP phone or softphone connected to Asterisk:
Dial an external number:
+12025550123
It should route through your bitcall-ip trunk — and start ringing. 🎉
🛡️ Securing IP Trunks
Since there’s no password, anyone spoofing your IP can use your trunk (and your money).
To protect it:
✅ Only allow calls to trusted destinations
✅ Use a firewall to allow SIP traffic only to gateway.bitcall.io
✅ Enable IP access control if using a GUI panel like FreePBX
Example (firewalld or iptables):
iptables -A INPUT -p udp -s 188.34.143.144 --dport 5060 -j ACCEPT
💡 Alternative: Use Bitcall’s IP Directly
If DNS fails or latency is critical, use:
host=188.34.143.144
Instead of gateway.bitcall.io
🧠 TL;DR Recap
✅ Add your public IP to Bitcall panel
✅ Configure sip.conf with no username/password
✅ Add outbound rules in extensions.conf
✅ Reload and start calling
❗ Use firewall rules to secure it!