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Flexible channels
Increase or reduce concurrent call capacity as demand changes.
NestDial SIP Trunking
Modern voice connectivity for the phone system you already trust.
Connect your existing PBX to scalable internet-based calling with flexible capacity and resilient routing.
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Business continuity
Use failover routes to keep critical calls moving.
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PBX compatibility
Connect common IP-PBX environments with guided onboarding.
How it works
A clear path to modern calling.
1. Review
We understand your PBX, numbers, volume and resilience needs.
2. Connect
Your team receives configuration details and setup support.
3. Launch
We test inbound, outbound and failover before moving traffic.
Why SIP trunking
Keep the PBX. Improve the connection.
SIP trunking can replace or complement traditional voice lines by carrying business calls over an IP connection. It is useful for organisations that already have a capable PBX but need more flexible channel capacity, centralised routing or continuity options.
Channel capacityScalable
Inbound and outboundSupported
Failover routingPlanned
Technical onboardingGuided
SIP questions
Frequently asked questions
What equipment do we need?
You need a compatible IP-PBX or session border environment and reliable network connectivity. NestDial will review the technical setup before activation.
How many simultaneous calls can we support?
Capacity is planned around concurrent channels. The right number depends on peak call traffic, staffing and continuity requirements.
Can SIP trunking provide backup routing?
Planned failover options can redirect calls to approved alternate destinations when the primary route is unavailable.
Is SIP trunking the same as a cloud phone system?
No. SIP trunking connects a compatible existing PBX, while a cloud phone system also provides the user, routing and administration platform.
Technical capabilities
Plan the trunk around traffic and resilience.
A useful SIP design considers peak calls, routing, security and what should happen during disruption.
Inbound and outbound routes
Define approved destinations and caller identities, including references such as +977 98XXXXXXXX.
Concurrent channels
Size capacity around the busiest realistic period and future growth.
Failover destinations
Plan alternate endpoints or an approved reference such as +977 97XXXXXXXX.
Codec alignment
Review compatible voice codecs and network conditions for dependable quality.
Security controls
Use appropriate authentication, access restrictions and encrypted transport where supported.
Call detail records
Keep metadata for troubleshooting, usage review and operational reporting.