Engineering capabilities
- Custom dial plan logic
- AMI / ARI integration
- Codec and transcoding strategy
- Queue and IVR design
VoIP
Asterisk
Open-source telephony engine used where dial plan logic, integrations and call flow behaviour need to be fully controlled.
Security & hardening
- TLS signalling and SRTP media
- Fail2ban and source restriction
- Outbound route limits against toll fraud
Asterisk reference architecture
Asterisk sits behind a hardened border, with dial plan logic and integrations kept in version-controlled configuration rather than ad-hoc GUI changes.
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01
Carrier & PSTN
- Primary SIP trunk
- Backup SIP trunk
- GSM / PRI gateway
Registration or IP authentication per trunk, with failover routes and CLI presentation rules.
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02
Border
- Edge firewall
- SIP ALG disabled
- Fail2ban / rate limits
Only carrier and known extension ranges reach 5060/5061; TLS for signalling, SRTP for media.
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03
Asterisk core
- PJSIP endpoints
- Dial plan contexts
- Queues & IVR
- AMI / ARI
Contexts separate internal, outbound and restricted classes; ARI/AMI exposed only to the application host.
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04
Data & integrations
- CDR database
- Recording storage
- CRM connector
CDR and recordings written to a separate volume with retention and access controls.
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05
Endpoints
- Desk phones
- Softphones
- Remote agents
Provisioning templates per device model; remote users routed via VPN or SBC rather than open SIP.
Frequently asked questions
When is raw Asterisk a better fit than FreePBX?
When call flow logic is genuinely custom u2014 routing driven by external data, complex integrations over ARI, or logic that must live in code review and version control. If the requirement is standard extensions, queues and IVR, a managed interface is cheaper to operate.
Which SIP stack should new deployments use?
PJSIP. It is the maintained channel driver and handles multiple contacts per endpoint, per-endpoint transports and TLS more predictably than the legacy chan_sip driver.
How do you stop toll fraud on an Asterisk box?
Strong per-device secrets, restricted outbound contexts, no default 'allow all' dial patterns, source-restricted SIP ports, fail2ban on registration failures, and spend/concurrency limits agreed with the carrier.
Can Asterisk run in a virtual machine?
Yes, with pinned CPU resources and attention to timing. Oversubscribed hosts cause jitter and audio artefacts, so voice VMs get reserved compute and are kept off noisy storage tiers.
Scope the work before you commit budget
Send the environment details and get an engineered proposal with assumptions, risks and a rollback path.