Next-Gen SIP Calling & Global DID Platform
Empower your enterprise with ultra-low latency SIP calling, automated DID management, local & international virtual numbers, and carrier-grade voice interconnects engineered for high-concurrency operations.
SIP Trunking & DID Node
Real-time CLI delivery, G.711 / G.729 HD codecs & automatic failover routing active.
Intelligent DID Management Platform
Take complete command of your phone number inventory. Provision local DIDs, toll-free lines, and international virtual numbers with instant dynamic routing.
Major City Area Codes
Instant allocation of clean DID blocks in Karachi (021), Lahore (042), Islamabad (051), Faisalabad (041), and Multan (061).
- Direct Carrier Interconnect (Zong 4G / Jazz)
- True CLI Pass-through for high pickup rates
Dynamic Call Routing
Route incoming DID calls dynamically based on time of day, caller location, agent availability, or IVR selections.
- Time-based and geo-location routing rules
- Real-time webhook and API trigger support
PBX Extension Mapping
Map individual DIDs directly to specific agents, departments, or Vicidial / Asterisk extensions with zero latency.
- Zero-touch SIP header customization
- Single or multi-tenant PBX support
Carrier-Grade SIP Infrastructure
Designed from the ground up to replace outdated E1/PRI circuits with high-performance, flexible SIP voice interconnects.
Ultra-High Concurrency
Scale from 10 to 10,000+ concurrent channels seamlessly. Perfect for large call centers, voice broadcast campaigns, and enterprise PBX environments.
Low-Latency Direct Interconnect
Direct fiber interconnects with top tier-1 carriers ensure sub-30ms latency, zero packet loss, and jitter-free voice performance.
Flexible Codec Negotiation
Full support for G.711a/u (uncompressed HD audio), G.729 (high efficiency low bandwidth), Opus, and GSM audio codecs.
TLS & SRTP Encryption
End-to-end voice encryption protecting customer conversations and signaling against eavesdropping and unauthorized interception.
Real-Time CDR & Analytics
Monitor call completion rates (ASR), average duration (ACD), MOS scores, and channel utilization live from a centralized dashboard.
Multi-Data Center Redundancy
Geographically distributed SIP proxies with automated IP failover guarantee uninterrupted uptime even during network outages.
Connect Any PBX, Dialer, or Cloud System
Our SIP trunks plug effortlessly into open-source platforms, enterprise hardware, and modern WebRTC applications with standard SIP digest authentication or IP address whitelisting.
Plug-and-Play Setup
Our technical support engineering team provides ready-made SIP context configurations for Asterisk, Vicidial dialplan strings, and 3CX trunk templates to get you live within hours.
4 Steps to Launch Your SIP Trunk
Requirements Audit
We analyze your concurrent call volume, target destinations, and existing PBX or dialer software.
DID Selection
Choose your city DID numbers (021, 042, 051), 0800 toll-free lines, or international virtual numbers.
SIP Interconnect
Provision SIP credentials or IP whitelist peering, test codec negotiation, and verify CLI delivery.
Go Live & Monitor
Switch voice traffic to Akyservices SIP backbone with 24/7 proactive NOC monitoring & SLA.
Everything About SIP Calling & DIDs
Ready to Upgrade to High-Concurrency SIP Calling?
Get in touch with our voice engineering team for free trial SIP trunk testing, custom DID allocations, and seamless PBX migration.