Frequently Asked Questions

Our direct interconnects with master Turkish operators and Tier-1 gateways allow us to guarantee average Post-Dial Delay (PDD) rates of less than 1.8 seconds. This keeps voice completion paths crisp and post-dial latencies short.

We support standard SIP (RFC3261) connections with audio codecs including G.711a, G.711u, and bandwidth-optimized G.729. Connections can be structured using Dynamic IP, static IP Authentication, or Secure SIP Digest (username/password) verification.

Yes, our SIP trunks are natively certified to connect with all primary telephone switchboards including Asterisk, FreePBX, Cisco Unified CallManager, Avaya Aura, 3CX, and Grandstream PBX nodes. Step-by-step configurations can be found inside our Partner Portal.

A predictive dialer is an automated pacing system that dials lists ahead of time. Our custom pacing algorithm tracks historical connection rates, average talk durations, and real-time agent availability to calculate the exact number of outgoing calls required to ensure agents transition smoothly into live talks with zero idle times.

Predictive mode: Dials ahead mathematically based on connection metrics. Best for larger teams (10+ agents) to handle drop risks.
Progressive mode: Dials only when an agent is verified free, eliminating dropped calls entirely. Best for high-value CRM leads.
Power mode: Dials a fixed ratio of calls per free agent (e.g., 3 lines per agent) continuously, maximizing database dialing speed.

Our advanced Digital Signal Processing (DSP) frequency scanner analyzes the initial voice pattern within 1.2 seconds of call pick-up. It accurately identifies long standard carrier greeting tones (which point to an answering machine) and silently drops the line before routing it to agents, maintaining high queue efficiencies.

Absolutely. We provide complete cryptographic shielding for signaling via TLS (Transport Layer Security) and encrypt media pathways with SRTP (Secure Real-time Transport Protocol). This ensures that voice packets cannot be intercepted or recorded.

Yes, our SMS HTTP API supports low-latency routing channels designed specifically for one-time passwords (OTP), two-factor authentication (2FA), and critical operational alerts. We support custom alphanumeric Sender IDs (Sender Header Masking) in all compliant regions.

To provide the absolute lowest rates for high-volume carrier and call center traffic, we negotiate real-time wholesale price points based on volume tiers. Instead of charging high static rates, our carrier trading desk prepares tailored rate cards matching your exact target destinations and active monthly call minutes. Contact us today to receive a rate card!

You can instantly trigger a rate proposal by using our custom quotation forms on this website, or connect directly with our active Sales NOC at sales@averixtelecom.com. Trial connections can be provisioned with test credits within 15 minutes.