IP PBX and call routing
Extension plans, dial rules, SIP trunks, sites, softphones and endpoints organized around how teams actually communicate.
- IP PBX
- SIP trunks
- Number plans
Voice as critical infrastructure
Every call routed with purpose, context and control.
AstSer engineers IP telephony platforms that connect extensions, trunks, service queues, recordings and business systems without sacrificing voice quality or operational visibility.
SIP
Standards-based voice
Solution overview
Telephony that behaves like an integrated business system.
AstSer designs new voice environments and modernizes existing PBXs with extension plans, SIP trunks, number portability, IVR, queues, recordings and endpoint strategy. The architecture accounts for sites, remote users, emergency routing and business continuity.
Open components such as Asterisk, FreeSWITCH and Kamailio can be combined with supported endpoints, carriers and business applications. Security, quality of service and observability are treated as core architecture—not post-deployment add-ons.
Engineering first
Architecture, security and operability are designed together from the start.
Built to integrate
Open standards and documented interfaces protect your existing investments.
SIP
Open signaling
TLS
Encrypted transport
SRTP
Protected media
HA
Resilient design option
Technical capabilities
A voice platform from dial tone to business workflow.
Routing, media, security and integrations are engineered against the same service objectives.
IVR, queues and contact flows
Context-aware menus, service queues, skills, schedules, overflow rules, callbacks and integrations with customer data.
- IVR
- Queues
- APIs
Security and call evidence
Boundary controls, fraud-reduction policies, encrypted signaling/media, role-based recordings, retention and traceability.
- SBC controls
- TLS/SRTP
- Recording
Quality and resilience
QoS policy, redundant components, carrier strategy, call-detail analysis and monitoring for fast diagnosis of voice issues.
- QoS
- CDR
- Failover
Delivery architecture
Voice engineered across network, carrier and application boundaries.
The design follows a call end to end—from endpoint and LAN to call control, trunk, media, recording and the external network.
Dial-plan simulations, failover scenarios and voice-quality tests reduce surprises during porting and cutover.
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01
Discover
Map users, assets, traffic, dependencies, risks and measurable operating objectives.
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02
Engineer
Define capacity, security controls, integration points, resilience and acceptance criteria.
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03
Implement
Deploy in controlled stages with validation, documentation, knowledge transfer and rollback planning.
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04
Operate
Monitor service health, resolve incidents and evolve capacity with the operation.
Platforms and ecosystem
Open architecture with operational ownership.
AstSer selects components for the required scale, routing logic and support model instead of binding the design to a single appliance.
Flexible IP PBX
Asterisk or FreeSWITCH-based call control can support extensions, applications, IVR, media services and custom routing.
Scalable voice edge
Kamailio and standards-based edge patterns support routing, registration, policy, distribution and resilient SIP services.
Where it creates value
Voice workflows for distributed and service-driven operations.
The same platform can simplify everyday calls and protect the communications that cannot stop.
Multi-site organizations
Unified dialing, centralized policies and local continuity across offices, branches and remote users.
Service centers
Queues, schedules, recordings and business-data integration for measurable customer interactions.
Critical communications
Controlled routing, evidence and failover planning for emergency, security and public-service workflows.
Frequently asked questions
IP telephony and VoIP: key questions
Technical answers to the questions organizations ask before planning a project.
Can we keep our existing numbers and phones?
Numbers can often be ported or retained through the carrier, and compatible SIP endpoints may be reused. AstSer inventories numbers, trunks, devices, analog dependencies and emergency services before defining the migration path.
How does AstSer protect a VoIP environment?
Controls can include network segmentation, hardened administration, SIP edge policy, strong identity, dialing restrictions, rate and fraud monitoring, TLS/SRTP where supported, patching, backups and auditable access.
Can VoIP remain available if a site or carrier fails?
Resilience can combine redundant call control, multiple trunks or carriers, survivable site routing, alternate destinations and tested runbooks. The appropriate level depends on business impact, location and budget.
Can IP Telephony | VoIP integrate with our current environment?
Yes. AstSer starts with an inventory of the current environment and designs IP Telephony | VoIP around supported standards, APIs, identity, network and security policies. Integration and migration criteria are validated before production rollout.
Design the next step
Build IP Telephony | VoIP around your operation.
Bring your sites, numbers, call flows and continuity goals. AstSer will design a secure voice roadmap with a controlled migration path.