BULKSMSGLOBAL
Enterprise VoIP & SIP Trunking | BulkSMSGlobal

Next-Gen VoIP
Infrastructure

Crystal clear voice quality over the cloud. Our SIP Trunking and VoIP termination services connect your business to the world with 99.99% uptime and enterprise-grade security.

Cloud VoIP Technology

Why Businesses Trust Our SIP Network

Advanced features for high-volume calling environments.

  • G.711 & G.729 Codec Support: Optimized bandwidth for HD audio.
  • Geo-Redundant Servers: Distributed nodes across USA, Europe, and Asia.
  • White-Label Billing: Fully brandable billing portal for resellers.
  • TLS/SRTP Encryption: Secure your calls with military-grade encryption.
  • Dynamic CLI Routing: Display your local brand number globally.
  • Fraud Detection: Real-time AI monitoring for suspicious call patterns.
Connectivity Center

Build Your Own Call Center

Our VoIP API allows you to deploy a full-scale contact center in hours, not weeks.

99.9% PDD

Low post-dial delay for instant connections.

Scalable Port

Handle 1,000+ CPS (Calls Per Second).

asterisk_sip_trunk.conf Active SIP Config

// BulkSMSGlobal SIP Trunk Integration
// Configure your PBX with our Global Nodes

[BulkSMSGlobal-Outbound]
type=peer
host=sip.bulksmsglobal.com
username=YOUR_ACCOUNT_ID
secret=YOUR_SECURE_TOKEN
context=outbound-calls
disallow=all
allow=ulaw,alaw,g729
nat=yes
qualify=yes

// The logic routes your call through the lowest latency node automatically
exten => _X.,1,Dial(SIP/BulkSMSGlobal-Outbound/${EXTEN})
exten => _X.,2,Hangup()
                

Compatible with Asterisk, FreePBX, 3CX, and Vicidial.