Transparent “from” pricing for real engineering work
Clear starting points for diagnostics, audits, emergencies, installation and advanced voice work. Final pricing depends on scope, urgency, system complexity and access requirements.
Focused written review of one specific issue.
- One SIP / RTP / ASR problem
- Call example review
- Configuration notes
- Written findings
- Recommended next steps
Asterisk, FreeSWITCH, Kamailio, OpenSIPS, RTPengine or trunk review.
- Architecture review
- Security review
- Routing review
- SIP / RTP risk review
- Written report
For urgent incidents.
- SIP / RTP fault isolation
- Carrier troubleshooting
- Failed inbound / outbound calls
- RTP / no-audio issues
- Written follow-up
Urgent incidents outside normal hours.
- Evening / weekend response
- SIP / RTP fault isolation
- Carrier troubleshooting
- Written follow-up
For ongoing support.
- Written technical support
- Periodic review
- Incident guidance
- Architecture improvements
- SIP / security recommendations
What pricing depends on. Final pricing depends on scope, urgency, system complexity and access requirements. Zypager does not promise guaranteed fixes — voice problems vary, and some require carrier cooperation or live access under appropriate change control. No payment is taken upfront — once scope and final price are agreed by email, you’ll get a secure payment link by card or PayPal.
Installation & deployment
done for you · per systemFixed “from” prices to install and configure a single system on your server — or combine several into a full build. Tick the ones you need and see the running total below.
Install Asterisk (pure)
£450Plain Asterisk from source with PJSIP — full control over dialplan and config.
fromInstall FreePBX
£550Asterisk plus the FreePBX web GUI.
fromInstall Issabel
£550Issabel unified-communications distro on Asterisk.
fromInstall Kamailio
£650Kamailio from the official repository with TLS / WebSocket modules.
fromInstall OpenSIPS
£650OpenSIPS with a routing skeleton for trunking.
fromInstall RTPengine
£550RTPengine with the in-kernel module for low-latency media.
fromKamailio + RTPengine
£950Wired together so media is relayed and one-way audio disappears.
fromOpenSIPS + RTPengine
£950OpenSIPS wired to RTPengine for reliable two-way audio.
fromAsterisk + Kamailio + RTPengine
£1,250The full stack — edge, media and application — integrated and documented.
fromKamailio High Availability
£2,000Floating edge IP, dispatcher failover and shared state across two nodes.
fromOpenSIPS High Availability
£2,000Highly available OpenSIPS edge with failover and shared state.
fromVoIP Edge Hardening
£750Firewall, rate limiting, TLS/SRTP and route restrictions.
fromAsterisk ARI development in Go
ARI · GoWritten review of an existing or planned ARI application.
- Call-control / state model review
- Event handling & recovery review
- Reliability & testing notes
- Recommended next steps
A baseline event-driven ARI application in Go.
- Go ARI application skeleton
- Connection & event handling
- Basic call-control flow
- Logging & error handling
- Run / test notes
A specific ARI workflow designed and implemented.
- IVR / recording / bridge flow
- State & error handling
- Recording or playback handling
- Testing & deployment notes
ARI integration for contact-centre call control.
- Agent / queue call-control flow
- Transfers & supervised handling
- Integration with backend services
- Reliability, logging & recovery
- Written implementation notes
Advanced voice applications & ASR
Vosk · offline ASRWritten review of whether Vosk suits a specific recording, transcription or voicebot workflow.
- Use-case & audio-quality review
- Codec / sample-rate considerations
- Language & model considerations
- Latency expectations
- Batch vs near-real-time options
- Recommended next steps
Baseline installation of a Vosk server for local speech-to-text.
- Vosk server deployment
- Model installation guidance
- WebSocket / gRPC endpoint notes
- Basic audio test
- Resource sizing notes
- Build / run documentation
Model choice, GPU/CPU sizing, scaling, custom vocabulary and Asterisk integration are quoted separately.
Integration for Asterisk recordings or ARI workflows using Vosk.
- Asterisk recording flow review
- ARI integration approach
- Audio format conversion approach
- Transcription storage design
- Written implementation notes
Live streaming, voicebot behaviour, diarisation and production deployment are quoted separately.
Pipeline design for converting recordings into searchable transcripts.
- Recording ingestion
- Job queue design
- Audio conversion
- ASR processing flow
- Transcript storage
- Privacy / redaction considerations
Architecture for near-real-time ASR used by voice agents or live assist.
- Indicative projects often start from £3,000+
- Streaming ASR design
- Latency considerations
- Asterisk ARI integration approach
- ASR / LLM / TTS flow
- Fallback & human handoff
Real-time ASR and voicebot projects depend on latency, audio quality, language, call volume and fallback requirements.
Common add-ons
Review of domain-specific terms, names, products, queues or call-centre phrases that may affect recognition quality.
Storing transcripts in PostgreSQL and making them searchable by interaction, recording, call ID, customer, date or agent.
Pipeline design for controlled summaries from call transcripts with privacy, redaction and human review considerations.
Future platform discovery
Discovery and architecture planning for future omnichannel contact-centre workflows, campaign automation, messaging channels, dialler strategy, chatbot integration and voice platform design.
This is not a ready-made product sale. Full platform delivery requires separate scoping, design and development.
Discuss discoveryPricing questions
You send one specific problem with a call example and any trace or configuration you can share. You receive written findings, likely root cause and recommended next steps — a focused, fixed starting price for a single issue.
For urgent, live incidents — failed inbound/outbound calls, RTP/no-audio and carrier problems — where the priority is fault isolation and restoring service. Scope and availability depend on access and complexity.
Ongoing written technical support, periodic review, incident guidance, architecture improvements and SIP/security recommendations. It suits teams that want a senior VoIP engineer on call without a full-time hire.
No payment is taken upfront. Once we've agreed the scope and final price by email, you'll receive a secure payment link — by card or PayPal — for that amount.
Not sure which service you need?
Send the symptoms and a call example. The right starting point becomes obvious from the trace.