What is Asterisk development?

Asterisk development is the process of building custom telephony applications using the Asterisk open-source PBX framework. CelloIP Technologies has delivered 100+ Asterisk deployments including multi-tenant IP PBX, AI-powered IVR, call center ACD, and FreePBX/FusionPBX customisations for enterprises across 30+ countries.

How much does custom Asterisk development cost?

Custom Asterisk development costs range from $15,000 for a single-site SME PBX to $120,000+ for enterprise multi-tenant call center platforms. CelloIP offers fixed-price milestones and dedicated Asterisk developer retainers from $25/hr.

How long does Asterisk PBX development take?

A basic Asterisk PBX deployment takes 4–8 weeks. A full call center platform with AGI, predictive dialler, and CRM integration takes 12–20 weeks. CelloIP uses agile sprints with weekly deliverables.

Asterisk Development

Expert Asterisk Development Services

CelloIP Technologies builds custom Asterisk PBX, IVR, AGI, and ARI solutions for enterprises, call centres, and telecom operators. 12+ years of Asterisk expertise — from single-office PBX to 500-agent omnichannel contact centres.

Asterisk 20/21PJSIPAGI/FastAGIARIFreePBXFusionPBXWebRTC
extensions.conf
[inbound-routes]
exten => _+1NXXNXXXXXX,1,NoOp(Inbound DID)
 same => n,Set(DID=${EXTEN})
 same => n,AGI(crm-lookup.php,${DID})
 same => n,GotoIf($[${CRM_FOUND}=1]?crm_route)
 same => n,Goto(ivr-main,s,1)
 same => n(crm_route),Dial(PJSIP/${AGENT},30)
 same => n,Hangup()

[ivr-main]
exten => s,1,Answer()
 same => n,Background(welcome-en)
 same => n,WaitExten(5)
exten => 1,1,Goto(queue-sales,s,1)
exten => 2,1,Goto(queue-support,s,1)
exten => 0,1,Dial(PJSIP/reception,20)

[queue-sales]
exten => s,1,Queue(sales,t,,60)
 same => n,Voicemail(sales@default,u)
 same => n,Hangup()
100+
Asterisk Deployments
12+
Years with Asterisk
500
Max Agents Deployed
99.97%
Average Uptime

Asterisk Development Services

From simple PBX setup to full omnichannel call centre development — every Asterisk use case covered.

Custom Asterisk PBX

End-to-end IP PBX deployment on Asterisk 20/21 with PJSIP. Extensions, ring groups, call queues, auto-attendant IVR, voicemail-to-email, call recording, CDR, and multi-tenant provisioning.

Asterisk 20/21PJSIPMulti-Tenant

IVR System Development

Multi-level IVR menus with Asterisk dialplan and AGI. Text-to-speech (TTS) integration, ASR (speech recognition), dynamic menu logic, CRM data lookups, and callback scheduling.

DialplanAGITTS/ASRCRM

AGI Script Development

Custom Asterisk Gateway Interface (AGI) scripts in PHP, Python, or Node.js. Business logic integration, database lookups, routing decisions, call scoring, and real-time CRM screen-pop.

FastAGIPHPPythonNode.js

ARI Application Development

Asterisk REST Interface (ARI) applications for full programmatic call control. WebSocket event-driven architecture, custom conferencing, call whispering, agent assist, and barge-in.

ARIWebSocketStasisREST

Call Centre on Asterisk

Full omnichannel contact centre built on Asterisk. ACD with skill-based routing, predictive dialler, real-time supervisor dashboard, agent state management, call recording, and analytics.

ACDPredictive DiallerQueueRealtime

FreePBX / FusionPBX Development

Custom FreePBX module development, FusionPBX multi-tenant configuration, Asterisk GUI customisation, and white-label PBX portal development for hosting providers.

FreePBXFusionPBXCustom Module

Asterisk Security Hardening

SIP digest auth enforcement, fail2ban configuration, AMI/ARI credential rotation, TLS/SRTP setup, context isolation, STIR/SHAKEN integration, and regular SIP trace security audits.

TLSSRTPfail2banSTIR/SHAKEN

Asterisk Migration & Upgrade

Migration from legacy Asterisk 11/13/16 to Asterisk 20/21 with chan_sip to PJSIP transition, dialplan audit, deprecated function replacement, and zero-downtime cutover.

Asterisk UpgradePJSIP Migrationchan_sip

Asterisk Version Guide: Which to Use in 2025?

Running an unsupported Asterisk version is a critical security risk. Here's the current support status.

VersionSupport StatusNotes
Asterisk 21 (LTS)✓ RecommendedSecurity supportedLatest LTS — recommended for new projects
Asterisk 20 (LTS)Full supportStable LTS — widely deployed in production
Asterisk 18 (LTS)Security onlyLegacy LTS — upgrade path available
Asterisk 16End of LifeEOL — must upgrade immediately
Asterisk 13End of LifeEOL — critical security risk

Our Asterisk Dialplan Expertise

CelloIP engineers have written production dialplans handling millions of daily calls across dozens of industries. We use Asterisk's full feature set — including the newer ARI-based Stasis application model for complex call flows that dialplan alone cannot handle cleanly.

Extension pattern matching (X, Z, N, [range])
GoTo, GotoIf, GotoIfTime branching
Macro and Gosub subroutines
Set / MixMonitor / Monitor
Queue() application with agent callback
Voicemail() and VoicemailMain()
AGI() and FastAGI() integration
Originate via AMI for click-to-call
CEL (Channel Event Logging) for analytics
PJSIP endpoint/transport configuration
SRTP media encryption config
ARI Stasis application entry point

Case Studies

US-Based BPO (500 Agents)

Challenge

Legacy Cisco UCM could not support predictive dialling or CRM integration without expensive licensing.

Solution

Migrated to Asterisk 20 with custom ARI-based predictive dialler, Salesforce CTI integration via SOAP API, real-time agent dashboard (WebSocket + React), and call recording with S3 archival.

Result

60% reduction in per-seat telephony cost, 34% improvement in agent connect rate, 3-second CRM screen-pop on inbound.

Middle East Telecom Operator

Challenge

Required a white-label hosted PBX platform for 200+ SME tenants with self-service provisioning.

Solution

Multi-tenant Asterisk cluster with FusionPBX as the management layer, custom REST API for reseller provisioning, Stripe billing integration, and CDR export in multiple formats.

Result

200+ tenants onboarded in 6 months, $0 per-seat licensing cost vs previous Avaya platform, 99.97% uptime in first year.

Asterisk Development: Technical Deep Dive

Asterisk is the world's most widely deployed open-source telephony platform, powering everything from small office PBX systems to large-scale contact centres handling tens of thousands of simultaneous calls. Originally created by Mark Spencer in 1999, Asterisk is now maintained by Sangoma Technologies and has an active global developer community.

Asterisk vs Commercial PBX: Why Open Source Wins at Scale

Commercial PBX vendors like Cisco, Avaya, and Mitel charge per-seat licensing fees that become prohibitive as organisations grow. A 200-seat Cisco deployment can easily cost $200,000–$400,000 in licensing alone, before hardware, professional services, or support contracts. Asterisk eliminates this licensing cost entirely. A well-architected Asterisk deployment on a $500/month cloud server can handle 500+ concurrent calls with features that match or exceed commercial alternatives.

When Should You Use Asterisk vs FreeSWITCH?

Asterisk excels as a feature-rich B2BUA (Back-to-Back User Agent) for PBX applications. Its dialplan system is extremely flexible for complex call routing, IVR trees, and feature logic. The AGI/ARI interfaces make CRM integration straightforward. Asterisk is the right choice for office PBX, call centre, IVR-heavy, and feature-rich enterprise communication deployments.

FreeSWITCH is better suited to high-concurrency softswitch applications — carrier-grade deployments handling 10,000+ simultaneous calls, large-scale conferencing, and scenarios requiring extensive transcoding. FreeSWITCH's event socket architecture (ESL) provides low-latency call control but has a steeper learning curve than Asterisk's dialplan model.

chan_sip to PJSIP Migration: What You Need to Know

The legacy chan_sip module was deprecated in Asterisk 17 and completely removed in Asterisk 21. Any deployment still running chan_sip is on an unsupported Asterisk version with unpatched security vulnerabilities. The PJSIP channel driver (based on the PJSIP library) provides better WebRTC support, native TLS/SRTP, IPv6, and improved multi-homed configuration. Migration requires converting sip.conf endpoint configurations to pjsip.conf format, updating dialplan PJSIP/ channel prefixes, and reconfiguring any AMI or AGI integrations that reference SIP channels directly. CelloIP provides fixed-price chan_sip to PJSIP migration services.

Asterisk Development FAQ

What is Asterisk development?

Asterisk development refers to the engineering work of building custom telephony solutions using the Asterisk open-source framework. This includes writing dialplans (extension routing logic), building AGI scripts for business logic integration, developing ARI applications for programmatic call control, configuring PJSIP endpoints, and integrating Asterisk with CRM, billing, and analytics systems.

What is the difference between Asterisk AGI and ARI?

AGI (Asterisk Gateway Interface) is a scripting interface that lets external programs control a call's dialplan execution. A FastAGI server written in PHP or Python receives dialplan execution events and returns instructions synchronously. ARI (Asterisk REST Interface) is a newer, more powerful interface that exposes Asterisk primitives via REST and WebSocket, allowing applications to have full programmatic control over channels, bridges, and playback — ideal for complex call flows, custom conferencing, and agent assist applications.

Should I use chan_sip or PJSIP in Asterisk?

PJSIP is the correct choice for all new Asterisk deployments. chan_sip was deprecated in Asterisk 17 and removed in Asterisk 21. PJSIP provides superior WebSocket transport for WebRTC, better TLS/SRTP support, IPv6 compatibility, and more flexible endpoint configuration. CelloIP migrates all legacy chan_sip deployments to PJSIP as part of our Asterisk upgrade service.

How many concurrent calls can Asterisk handle?

Asterisk scales to 3,000–5,000 concurrent calls on a well-provisioned server (16-core CPU, 32GB RAM) depending on transcoding load and media handling. Transcoding (G.711 to G.729, for example) is CPU-intensive and reduces capacity significantly. For call volumes above 5,000 concurrent sessions, we recommend a hybrid architecture: Asterisk for feature-rich application logic combined with OpenSIPS or Kamailio for high-volume SIP routing.

Can Asterisk integrate with Salesforce, HubSpot, or other CRMs?

Yes. CelloIP has built Asterisk-CRM integrations for Salesforce (REST API + CTI Toolkit), HubSpot (Webhook + Calling SDK), Zoho CRM, Microsoft Dynamics 365, and Freshdesk. Standard integration features include click-to-call, screen-pop on inbound, automatic call logging, call disposition codes, and contact lookup. Integration is typically built as a FastAGI or ARI application.

How long does a custom Asterisk PBX deployment take?

A standard Asterisk IP PBX for a 20–50 seat office (extensions, IVR, queues, call recording, voicemail) takes 2–4 weeks. A multi-tenant hosted PBX with billing and provisioning portal takes 8–16 weeks. A full call centre platform with predictive dialler and CRM integration takes 12–20 weeks. All timelines are confirmed in the architecture proposal after a free discovery session.

Flexible Engagement

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Hire an Asterisk Developer Today

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