SIP Trunking Service

Features & Capabilities

• SIP Trunk Lines: Virtual phone lines connecting your existing on-premise PBX to the public phone network over the internet.
• DID Number Provisioning: Direct inward dial numbers for every employee, department, or location — no new hardware required.
• Unlimited Calling Plans: Flat-rate unlimited US calling plans for predictable monthly costs — no per-minute surprises.
• Number Porting: Keep all your existing phone numbers. ATS Voice manages the port from start to finish — zero-downtime cutover.
• E911 Compliance: Accurate emergency 911 location service registered for every trunk line and DID number.
• Disaster Recovery Failover: Calls automatically reroute to mobile or backup numbers if your internet connection goes down.
• Fraud Protection: Real-time fraud monitoring detects call anomalies and suspicious international dialing patterns — automatic alerts.
• QoS Monitoring: Built-in call quality monitoring tracks latency, jitter, and packet loss for clear calls at all times.
• BYOT (Bring Your Own Trunk): Compatible with Avaya, Cisco, NEC, Grandstream, Mitel, and virtually any SIP-capable PBX.
• Least-Cost Routing: Automatically routes calls over the most cost-effective path — reduces per-minute costs on high-volume lines.
• Scalable Channels: Add or remove trunk channels instantly — unlike PRI circuits, SIP scales in single-channel increments.

Why ATS Voice

• Save 40–60% vs traditional POTS lines or PRI circuits
• Keep your existing PBX — no hardware replacement required
• PRI circuits at $400–$800/month replaced at $150–$300/month
• Free line cost analysis using your current phone bill
• ATS Voice has ported hundreds of numbers for East Tennessee businesses with zero reported downtime

Frequently Asked Questions

What is SIP Trunking?

SIP Trunking replaces traditional phone lines — whether POTS (Plain Old Telephone Service) copper lines or PRI (Primary Rate Interface) circuits — with virtual connections that carry voice calls over your existing internet connection using SIP (Session Initiation Protocol). Your on-premise PBX continues to operate exactly as it always has; only the connection to the public phone network changes. The result is a significant reduction in monthly line costs without requiring any changes to the phones, features, or call flows your staff already knows. ATS Voice has provided SIP Trunking to East Tennessee businesses for over a decade, replacing outdated line cards with modern, scalable virtual channels.

Will I need to replace my PBX?

No — SIP Trunking is specifically engineered to extend the life of your existing PBX investment. ATS Voice SIP trunks are compatible with all major PBX platforms including Avaya, Cisco, NEC, Mitel, Grandstream, Yealink, and most other SIP-capable systems. You keep the hardware your team already knows, the extensions stay the same, and your call flows remain unchanged. The only thing that changes is the line side of the PBX — analog or PRI circuits are replaced with SIP channels. ATS Voice performs compatibility verification for your specific PBX model before provisioning, so there are no surprises during cutover.

How much can I save?

Most businesses that switch from traditional POTS lines or PRI circuits to ATS Voice SIP Trunking save between 40% and 60% on their monthly phone line costs. The exact savings depend on your current provider, the number of lines or PRI channels you have, and your outbound call volume. PRI circuits commonly run $400–$800 per month for 23 channels; equivalent SIP capacity with ATS Voice typically costs $150–$300. Businesses with multiple POTS lines see similar proportional savings. ATS Voice provides a free line cost analysis using your current phone bill to calculate your specific projected savings before you make any commitment.

Can I keep my existing phone numbers?

Yes — all your existing business phone numbers port to ATS Voice SIP Trunking at no charge. We manage the porting process directly with your current carrier, and your service remains uninterrupted throughout the transition. Standard porting timelines for SIP are 7–14 business days for most carriers. During that period, your existing lines stay active so no calls are missed and no customers experience any disruption. ATS Voice has ported hundreds of numbers for East Tennessee businesses, including complete PRI DID blocks and multi-location number sets. Our local team handles each port directly rather than routing it through an offshore ticketing system.

What happens if my internet goes down?

ATS Voice SIP Trunking includes automatic disaster recovery failover on every trunk. If your internet connection becomes unavailable — whether from an ISP outage, equipment failure, or a weather event — the ATS Voice platform detects the loss of SIP registration within seconds and automatically reroutes all incoming calls to preconfigured backup destinations: mobile phones, an alternate office location, an answering service, or voicemail. This failover happens in the cloud, with no action required on your end. For East Tennessee businesses where connectivity can be affected by severe weather, this built-in resilience ensures customers can always reach you regardless of local conditions.

How many SIP channels do I need?

The rule of thumb is one SIP channel per simultaneous call — one channel handles one active conversation. Unlike PRI circuits that sell in fixed 23-channel increments, SIP channels are provisioned in any quantity you need. A 50-person office that analyzed actual call data typically peaks at 10–15 concurrent calls, meaning 15–18 channels provides comfortable capacity with overflow headroom. ATS Voice reviews your current call volume data — or your existing PRI/line count as a baseline — and recommends the right channel count. Channels can be added in minutes if call volume grows, and you only pay for what you use each month.

Is SIP Trunking secure?

Yes — ATS Voice SIP Trunking includes multiple layers of security on every account. All SIP signaling and RTP media streams are encrypted using TLS (Transport Layer Security) and SRTP (Secure Real-time Transport Protocol), preventing eavesdropping and man-in-the-middle attacks. Real-time fraud monitoring watches for call anomalies such as sudden international dialing spikes or off-hours call bursts, with automatic alerts and trunk suspension if suspicious activity is detected. Every trunk includes E911 compliance with registered service address for each DID, meeting FCC requirements. ATS Voice has maintained these security standards across its East Tennessee customer base for over 23 years.

How long does setup take?

New SIP trunks with new phone numbers can be provisioned and active within 1–2 business days. If you are porting existing numbers from your current carrier, that process adds 7–14 business days — during which your current lines remain fully operational so there is zero call interruption. The PBX-side configuration (updating your SIP trunk settings to point to ATS Voice) typically takes 1–2 hours for a certified technician and can be scheduled during off-hours to avoid any business disruption. ATS Voice provides PBX configuration support for all major platforms and has completed same-day cutovers for East Tennessee businesses with urgent migration timelines.

What internet speed do I need?

Each simultaneous SIP call requires approximately 100 Kbps of bandwidth (using the G.711 codec standard for business voice quality). This means a standard 50 Mbps business internet connection can theoretically support over 500 concurrent calls with bandwidth remaining for other traffic. In practice, a 20-channel SIP trunk uses only about 2 Mbps of your pipe. ATS Voice also recommends configuring Quality of Service (QoS) rules on your router to prioritize voice packets, preventing call quality degradation during periods of heavy data usage. Our East Tennessee support team provides QoS guidance for all common router and firewall platforms as part of the SIP Trunking onboarding process.

What support do you provide?

ATS Voice provides local East Tennessee support from real technicians who specialize in business telephony — not an outsourced call center staffed by generalists. Support is available Monday through Friday, 8 AM to 6 PM Eastern, with after-hours emergency coverage for critical trunk outages. When you call, you reach a person familiar with your specific PBX platform, your trunk configuration, and your business. This is backed by 23+ years in the telecom industry serving 650+ East Tennessee businesses. There are no national 1-800 queues, no automated tiering systems, and no overseas support desks — the same local team that provisioned and configured your SIP trunks handles every ongoing support need.