SIP Trunk Configuration

Connect Callab AI to your existing telephony infrastructure via SIP trunk.

CRITICAL: IP Whitelisting Required

User MUST whitelist Callab AI SIP IP in their SIP server/PBX:

Callab AI IPs:

• SIP Signaling: 104.197.91.206:5060
• RTP Audio: 35.202.135.16 (ports 20000-60000)

Add these IPs to:

• SIP server whitelist
• Firewall allow rules
• PBX trusted peers
• SIP trunk ACL

Without whitelisting, SIP trunk will NOT work.

SIP trunk connects Callab AI agents to existing telephony infrastructure:

• Use existing phone numbers
• Integration with on-premise PBX
• Enterprise telephony systems
• Regional/in-country calling

Add SIP Trunk

User can add SIP trunk:

1. Go to Phone Numbers
2. Click “Add Phone Number”
3. Select “SIP Trunk”
4. Configure settings
5. Save

Section 1: Phone Number & Concurrency

Phone Number:

• Format: E.164 ([phone])
• Appears as caller ID
• Must be owned by user
• Used for inbound/outbound

Nickname:

• Friendly identifier
• Example: “Main Office”, “Support Line”
• Optional but recommended

Concurrency:

• Max simultaneous calls
• Match carrier limits
• Don’t exceed capacity
• Based on plan tier

Section 2: SIP Configuration

E.164 Append (+):

• Auto-prepend ”+” to numbers
• Enable if carrier requires E.164
• Default: Disabled

Requires Registration:

• Enable for authentication
• Callab registers with SIP server
• Most carriers: Enabled

Registration Using Public IP:

• Use Callab’s public IP
• Enable if carrier requires
• Default: Enabled

Tech Prefix:

• Prefix before destination number
• Example: “9” for outbound
• Leave empty if not needed

Diversion Headers:

• Include SIP diversion headers
• For call forwarding info
• Default: Disabled

User can configure SIP trunk using two authentication methods (see below).

SIP Trunk Authentication Types

User must choose between two SIP trunk authentication methods:

Register-Based SIP Trunk (Username/Password)

How It Works:

• Callab registers with your SIP server
• Uses username and password authentication
• SIP REGISTER messages sent periodically
• Server authenticates Callab AI

When to Use:

• SIP provider requires authentication
• Using hosted SIP service
• Carrier provides credentials
• Multiple customers on same IP

Configuration:

• Requires Registration: Enabled
• Provide SIP username
• Provide SIP password
• Gateway hostname/IP
• Callab sends REGISTER to authenticate

Example Providers:

• Most cloud SIP providers
• VoIP.ms
• Flowroute
• Bandwidth.com

IP-Authenticated SIP Trunk (INVITE-Based)

How It Works:

• No registration required
• Authentication by IP address
• Direct SIP INVITE messages
• Server whitelists Callab AI IP

When to Use:

• On-premise PBX systems
• IP-based authentication
• Internal corporate systems
• Direct SIP trunk providers

Configuration:

• Requires Registration: Disabled
• No username/password needed
• Gateway hostname/IP only
• Your server MUST whitelist: 104.197.91.206

Example Systems:

• Asterisk/FreePBX with IP authentication
• FreeSWITCH with ACL
• 3CX with IP-based trunk
• Cisco CUCM

IP-Authenticated Trunks:

When using IP authentication (Registration: Disabled):

• Your SIP server MUST whitelist 104.197.91.206:5060
• Your firewall MUST allow 35.202.135.16 (RTP ports 20000-60000)
• No username/password needed
• Authentication is by source IP only

Section 3: Gateways

Gateway IP/Hostname:

• SIP server address
• IP: 192.168.1.100
• Hostname: sip.carrier.com
• Port optional (default 5060)

Direction:

• Inbound: Receive only
• Outbound: Make only
• Both: Bidirectional (most common)

Protocol:

• UDP: Standard (recommended)
• TCP: Firewall compatibility
• TLS: Encrypted/secure

SIP OPTIONS Pings:

• Keepalive mechanism
• Detects failures
• Default: Disabled

User can add multiple gateways for failover, load balancing, or regional routing.

Internal Extensions

User can add internal extensions for routing to specific departments or users:

Extension Format:

• Use numeric extensions
• Recommended length: 5 digits
• Example: 10001, 10002, 10100

How It Works:

• User dials extension during call
• PBX routes to specific destination
• AI can transfer to extension
• Works with transfer_call tool

Configuration Examples:

Sales Extension:

• Extension: 10001
• Routes to: Sales team queue
• AI transfers: “Let me transfer you to sales”

Support Extension:

• Extension: 10002
• Routes to: Support team
• AI transfers: “Transferring to support”

Specific User:

• Extension: 10100
• Routes to: John’s direct line
• AI transfers: “Connecting you to John”

Best Practices:

• Use 5-digit extensions (recommended)
• Consistent numbering scheme
• Document extension mapping
• Test all extensions before production
• Avoid conflicts with phone numbers

Extension Requirements:

When using extensions:

• Configure in your PBX dial plan
• Test routing before deployment
• Ensure extensions are reachable from Callab trunk
• Document all extension assignments

Audio Codecs

Callab AI supports the following audio codecs:

Supported Codecs:

• PCMU (G.711 μ-law) - Standard, best compatibility
• PCMA (G.711 A-law) - Standard, international
• G722 - Wideband, better quality
• OPUS - High quality, low bandwidth

Codec Selection:

• Callab negotiates codec with SIP server
• Uses highest quality both support
• Falls back to PCMU if needed
• Configure allowed codecs in PBX

Best Practices:

• Enable all codecs for compatibility
• Prefer G722 or OPUS for quality
• PCMU/PCMA for maximum compatibility
• Test codec negotiation before production

Configuration Examples:

Asterisk:

# /etc/asterisk/pjsip.conf or sip.conf
disallow=all
allow=opus
allow=g722
allow=ulaw
allow=alaw

FreeSWITCH:

<param name="codec-prefs" value="OPUS,G722,PCMU,PCMA"/>

3CX:

• Go to SIP Trunk settings
• Select Codecs tab
• Enable: OPUS, G722, PCMU, PCMA
• Save configuration

IP Whitelisting (MANDATORY)

User MUST whitelist both Callab AI IPs:

• SIP Signaling: 104.197.91.206:5060
• RTP Audio: 35.202.135.16 (ports 20000-60000)

Add to:

• SIP server firewall
• PBX ACL
• Network firewall
• Security groups

This is REQUIRED. Calls fail without whitelisting.

Whitelisting Instructions

Asterisk/FreePBX (PJSIP):

# /etc/asterisk/pjsip.conf
[callab-trunk]
type=endpoint
context=from-callab
disallow=all
allow=ulaw,alaw,g722,opus
aors=callab-trunk

[callab-trunk]
type=aor
contact=sip:104.197.91.206:5060

[callab-trunk]
type=identify
endpoint=callab-trunk
match=104.197.91.206
match=35.202.135.16

Asterisk/FreePBX (SIP):

# /etc/asterisk/sip.conf
[callab-trunk]
type=peer
host=104.197.91.206
port=5060
context=from-callab
disallow=all
allow=ulaw,alaw,g722,opus
permit=104.197.91.206/32
permit=35.202.135.16/32

FreeSWITCH:

<!-- /etc/freeswitch/sip_profiles/external/callab.xml -->
<include>
  <gateway name="callab-trunk">
    <param name="proxy" value="104.197.91.206:5060"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="codec-prefs" value="PCMU,PCMA,G722,OPUS"/>
  </gateway>
</include>

<!-- /etc/freeswitch/autoload_configs/acl.conf.xml -->
<list name="callab" default="deny">
  <node type="allow" cidr="104.197.91.206/32"/>
  <node type="allow" cidr="35.202.135.16/32"/>
</list>

3CX:

1. Go to SIP Trunks
2. Edit trunk
3. Add 104.197.91.206 to allowed IPs
4. Add 35.202.135.16 to allowed IPs
5. Set codecs: PCMU, PCMA, G722, OPUS
6. Save

Firewall:

# Allow SIP signaling from Callab
iptables -A INPUT -p udp -s 104.197.91.206 --dport 5060 -j ACCEPT

# Allow RTP audio from Callab
iptables -A INPUT -p udp -s 35.202.135.16 --dport 20000:60000 -j ACCEPT

REMINDER: Whitelist Both IPs

Most common error is forgetting to whitelist these IPs.

Check:

1. SIP IP 104.197.91.206:5060 whitelisted?
2. RTP IP 35.202.135.16 whitelisted?
3. Port 5060 open (UDP)?
4. Firewall rules correct?
5. RTP ports 20000-60000 open?

SIP Trunk Communication Flow

Standard Flow

Key Points:

• SIP signaling: 104.197.91.206:5060
• Audio RTP: 35.202.135.16 (ports 20000-60000)
• Both IPs must be whitelisted
• UDP recommended
• Supported codecs: PCMU, PCMA, G722, OPUS

Regional/Enterprise Flow

Enterprise Plan Only:

Regional routing uses Callab’s in-country servers for:

• Lower latency (< 50ms)
• Local compliance
• Better quality
• Data residency

Available: Saudi Arabia, UAE

Enterprise Features:

• Traffic stays in-country
• Meets local regulations
• Regional IP whitelisting
• Contact sales to enable

Testing

Test Outbound:

1. Configure SIP trunk
2. Assign to agent
3. Make test call
4. Verify connection
5. Check audio quality
6. Confirm caller ID

Test Inbound:

1. Call SIP trunk number
2. Verify routing
3. Confirm agent answers
4. Test conversation
5. Check audio both ways

Common Issues:

Issue	Cause	Fix
No connection	SIP IP not whitelisted	Add 104.197.91.206:5060
No audio	RTP IP blocked	Whitelist 35.202.135.16:20000-60000
Auth failed	Wrong credentials	Check username/password
One-way audio	NAT/firewall	Configure NAT, check RTP
Wrong caller ID	E.164 setting	Check E.164 Append setting
Codec mismatch	Unsupported codec	Enable PCMU/PCMA/G722/OPUS

Troubleshooting

Calls don’t connect:

• Is SIP IP 104.197.91.206:5060 whitelisted?
• Is port 5060 open (UDP)?
• Is gateway IP/hostname correct?
• Are credentials correct (if using registration)?
• Is firewall blocking SIP traffic?

One-way audio:

• Is RTP IP 35.202.135.16 whitelisted?
• Are RTP ports 20000-60000 open (UDP)?
• Is NAT configured correctly?
• Is firewall blocking RTP traffic?
• Are codecs compatible (PCMU/PCMA/G722/OPUS)?

Registration fails:

• Gateway IP correct?
• Credentials correct?
• “Requires Registration” enabled?
• Server allows 104.197.91.206?

Best Practices

Configuration:

• Always whitelist both IPs: 104.197.91.206 and 35.202.135.16
• Configure allowed codecs: PCMU, PCMA, G722, OPUS
• Set proper concurrency limits
• Test before production deployment
• Document all extension mappings

Security:

• Whitelist only Callab IPs (no other sources)
• Use strong SIP passwords (if using registration)
• Enable TLS if supported by infrastructure
• Monitor for unauthorized access attempts
• Log SIP traffic for troubleshooting

Performance:

• Match concurrency to capacity
• Use UDP for performance
• Configure QoS
• Monitor quality metrics
• Use regional for latency

Reliability:

• Configure backup gateways
• Enable keepalive
• Monitor health
• Set up alerts
• Test failover

Common Configurations

Register-Based (Username/Password):

Phone: [phone]
Concurrency: 10
E.164 Append: Disabled (adjust per carrier)
Registration: Enabled
Username: your_sip_username
Password: your_sip_password
Gateway: sip.carrier.com:5060
Direction: Both
Protocol: UDP
Codecs: PCMU, PCMA, G722, OPUS
Whitelist: 104.197.91.206:5060, 35.202.135.16:20000-60000

IP-Authenticated (INVITE-Based):

Phone: [phone]
Concurrency: 10
E.164 Append: Disabled
Registration: Disabled
Gateway: 192.168.1.100:5060
Direction: Both
Protocol: UDP
Codecs: PCMU, PCMA, G722, OPUS
Whitelist: 104.197.91.206:5060, 35.202.135.16:20000-60000
Note: Your server MUST whitelist Callab IPs

Enterprise with Failover:

Phone: [phone]
Concurrency: 100
Primary: sip-primary.carrier.com:5060
Backup: sip-backup.carrier.com:5060
Registration: Enabled
Codecs: PCMU, PCMA, G722, OPUS
Whitelist: 104.197.91.206:5060, 35.202.135.16:20000-60000

Support

Help:

• Email: [e-mail]
• Include: Configuration details
• Include: SIP error logs
• Include: Test results

FINAL REMINDER: Whitelist Both Callab AI IPs

Required IPs:

• SIP Signaling: 104.197.91.206:5060
• RTP Audio: 35.202.135.16 (ports 20000-60000)

Required in:

• SIP server config (allow both IPs)
• Firewall rules (ports 5060 and 20000-60000)
• PBX settings (whitelist/ACL)
• Network security groups

Supported Codecs: PCMU, PCMA, G722, OPUS

Calls will fail without whitelisting both IPs.