IP PBX

Simgenet SMG PBX Server

The Simgenet SMG PBX Server is an in-house developed enterprise phone system that brings all of your organization's voice communication together in one modern platform. Extensions, external call trunks, voice menus, call queues and conferencing are all covered on a single system; if you wish, an AI-powered voice assistant can answer calls automatically. Your calls are protected with encryption and secured against fraud and toll-abuse risks. It grows from a small office to an organization with tens of thousands of users; installation, maintenance and support are provided from Türkiye with Simgenet assurance.

Simgenet SMG PBX Server


The Simgenet SMG PBX Server is a modern enterprise phone system that lets you manage all of your organization's voice communication from a single place. From your employees' extensions to the external trunks that connect you to your operator, from voice menus and call queues to conference calls, it brings together on one platform every telephony feature an organization needs. It works with IP phones of different brands and models; as your organization grows, you can easily add new users and features. The product is entirely developed in Türkiye; installation, maintenance and technical support are provided with Simgenet assurance.

Embedded AI Voice Assistant

The most distinctive feature of the product is the AI voice assistant embedded in the panel. With components that convert speech to text (STT), interpret it with a large language model, and turn text back into speech (TTS), it provides IVR automation, call summarization, automatic conference joining and an organization-specific trainable knowledge base. The AI Studio module allows these scenarios to be managed and trained from the web interface.

Announcement System and How It Works

The announcement (paging) system of Simgenet SMG PBX Server is a powerful module that enables live, one-way voice announcements across an entire facility. Announcement points are organized geographically and functionally into a hierarchy of region, group and terminal (announcement station). When an operator starts an announcement, all speaker and terminal points in the selected group are automatically joined to an Asterisk ConfBridge conference room; when the operator speaks, every point in the group listens at the same time. Terminal microphones are muted by default, keeping the announcement one-way and clear.

For high capacity, announcements are not squeezed into a single room; groups are distributed across multiple conference rooms to reach thousands of points reliably, and more than 2,000 concurrent points have been verified in Simgenet's R&D tests. Live participation state is kept in real time on Redis, while region, group and permission configuration is managed in the database. Thanks to role-based authorization, the global administrator manages all regions, a regional administrator manages only their own region, and the control-room operator only makes announcements. Announcement terminals run dedicated station software with acoustic echo cancellation (AEC) to prevent feedback problems in loud environments.

Security — 7-Layer Defense

The security of Simgenet SMG PBX Server is designed with a defense-in-depth approach. The Kamailio-based SIP security gateway repels external attacks, while intrusion prevention, firewall, geographic filtering, threat feed and toll-fraud protection prevent unauthorized access and abuse. All voice communication is encrypted end to end with SIP/TLS and SRTP, the system is protected by ClamAV antivirus and rootkit scanning, and every action is recorded in the audit log. Enterprise authentication is centrally handled through RADIUS, LDAP and TACACS+, and authorization is handled by a role-based permission matrix.

Management and Ease of Use

The management interface is comprehensive yet simple and user-friendly. Thanks to its Excel-like matrix table structure, complex configurations (user and permission, line and profile, rule and condition mappings) are managed easily on a single screen. All operations are carried out from one panel through the dashboard, lines and users, routing, paging, calls, monitoring and reports, security, AI Studio and storage modules.

Scalability and Integration

The product uses the same codebase from the single-server SMG PBX Edge edition to the distributed SMG PBX Server edition designed for large enterprises. With its Kamailio and multi-node Asterisk cluster architecture, it scales from 100 users to 100,000 users and provides high availability with N+1 redundancy. Call recordings are archived on the Simgenet SMG Unified Storage server (S3); it integrates with ISP SIP lines, gateways and CRM systems.

Engine / Platform
Call EngineAsterisk 22 LTS (B2BUA + media bridge)
SIP Security GatewayKamailio 5.6.3+ (external SIP gateway, port 5061 TLS)
Operating SystemSimgenet appliance (Debian Linux based)
RoleProtocol/codec converter — clients with different encryption/transport/codec interoperate simultaneously
Product Family (single codebase, EDITION flag)
SMG PBX EdgeSmall-medium business (5–100 extensions)
SMG PBX ServerMedium-large enterprise (100–10,000+ extensions)
SMG UC ApplianceProxmox + firewall + Edge VM (unified communications)
SMG AI ModuleGPU AI add-on (voice assistant)
SIP / Signaling
SIPRFC 3261 · UDP / TCP / TLS · PJSIP
TLS PortsKamailio 5061 (external) · Asterisk 5062 (internal)
MethodsREGISTER, INVITE, REFER, SUBSCRIBE/NOTIFY, MESSAGE
RealtimePJSIP realtime (DB-backed endpoints)
Media / Encryption
MediaRTP / SRTP
SRTP KeyingSDES (RFC 4568) · DTLS-SRTP (RFC 5764, WebRTC)
TLSTLS 1.2/1.3 · ECDHE-RSA-AES256-GCM-SHA384
TranscodingCodec conversion (e.g. Opus ↔ G.711)
Client Profiles (concurrent)
WebRTCWSS (TLS/WebSocket) · DTLS-SRTP (mandatory) · Opus/G722/G711 — browser softphone
Secure-IPTLS 5061 · SRTP (SDES) · Cisco/Yealink/Grandstream (TLS)
Classic-IPUDP/TCP · classic IP phones
AutoAutomatic profile detection (AMI REGISTER listener)
Audio Codecs
WidebandOpus · G.722 (HD voice)
NarrowbandG.711 (A-law / µ-law) · G.729
NAT Traversal
MethodsSTUN / TURN / ICE
Telephony Features
ExtensionsPJSIP / SIP · user group + profile
TrunksSIP / PJSIP · TLS + SRTP · DID / inbound-outbound routing
IVRMulti-level voice menu · time conditions · follow-me
Call Queue (ACD)Queue / call distribution · ring groups
ConferenceConfBridge · AI auto-join · room codes
PagingLive + scheduled + triggered group paging · high-capacity listener (2,000+ concurrent, R&D-proven) · zone/tenant filtered · text-to-speech (TTS) announcements · priority/collision handling
Call TransferREFER (RFC 3515) · attended / blind
BLFSUBSCRIBE/NOTIFY (dialog-info) · phone LED
VoicemailVoicemail · MWI (RFC 3842) · email notification
ExtraPark · Pickup · DND · Dial Plan
AI Voice Assistant (AI Studio)
EngineSTT (speech-to-text) + LLM + TTS (text-to-speech)
CapabilitiesIVR automation · call summary · automatic conference join
Knowledge BaseOrganization-trainable RAG knowledge base
StudioWeb-based scenario/flow/training management
Security — 7-Layer Defense-in-Depth
Secure ModeMandatory encryption mode (unencrypted registration rejected)
SIP Security GatewayKamailio — topology hiding · rate limiting · SIP filtering
Intrusion Prevention (IPS)Real-time attack detection + auto-blocking (fail2ban class)
Toll-Fraud ProtectionInternational/abnormal call detection + blocking
AAA (Enterprise Identity)RADIUS / LDAP / TACACS+ central identity
Threat FeedLive threat / IP-reputation feed
FirewallHost firewall + GeoIP filtering
AntivirusClamAV malware scanning
Rootkit ScanSystem integrity / rootkit checks
TLS Certificate ManagementRoot CA + server certificate · auto-renewal
Authorization / AuditRBAC permission matrix + audit log · two-layer password (web admin given, OS root withheld)
CDR / Monitoring / Reporting
CDR / CELCall-detail + event records
RetentionMonthly partitioning + 10-year retention (automatic maintenance)
Monitoring & ReportsLive dashboard · call statistics · extension event logs (CSV/TXT)
Recording
TargetsLocal + SFTP / NFS / SMB / S3 · retention
IntegrationArchive with Simgenet SMG Storage (S3)
Scalability and Capacity
Node Capacity~2,000 concurrent calls / node (8 vCPU / 23 GB, proven)
ScaleFrom 100 to 100,000 users — Kamailio + multi-Asterisk cluster
High AvailabilityDistributed, N+1 redundant, automatic failover
FederationZero-touch SMG PBX merge (across branches)
Web Management Panel (Modules)
Management ExperienceComprehensive yet simple — Excel-style matrix tables (user × permission, line × profile, rule × condition); powerful, user-friendly single panel
Main ModulesDashboard · Lines & Users · Routing · Paging · Calls · Monitoring & Reports · Security · AI Studio · Storage
SystemLicense · System Health · Users · Network · PBX Settings · PJSIP+Conference Tunables · Service On/Off · Maintenance & Updates · Config Backup · Logs · Permission Matrix
MonitoringSNMP + custom MIB / NMS integration
Compliance and Deployment
Legal ComplianceBTK · KVKK
DeploymentISO / .deb + APT repo
BackupConfig backup + disaster recovery (automatic scheduled)
IntegrationPhone / ISP / gateway / CRM · SMG Storage
Environmental
Operating Temperature-20°C ~ +60°C
Storage Temperature-40°C ~ +70°C
Relative Humidity10% ~ 95% (non-condensing)
EMC and Compliance
EMC EmissionTS EN 55032:2015+A11:2020 (Class A)
EMC ImmunityTS EN 55035:2017+A11:2020
ESD (IEC 61000-4-2)Level 3
Radiated RF (IEC 61000-4-3)Level 3
EFT/Burst (IEC 61000-4-4)Level 3
Surge (IEC 61000-4-5)Level 3
Conducted RF (IEC 61000-4-6)Level 3
Magnetic Field (IEC 61000-4-8)Level 3
Voltage Dips (IEC 61000-4-11)Level 3
HarmonicsTS EN IEC 61000-3-2:2019+A1:2021
FlickerTS EN 61000-3-3:2013+A2:2021
Safety (LVD)TS EN 62368-1:2020+A11:2020
RoHSTS EN IEC 63000:2018
EU Directives2014/30/EU (EMC), 2014/35/EU (LVD), 2011/65/EU+2015/863/EU (RoHS)
Target Segment and Use Cases
Target SegmentEnterprise / campus · call center · multi-branch business · hotel/hospital/factory
Use CasesEnterprise IP telephony · call center (IVR/queue) · conferencing + AI assistant · branch-HQ SIP · paging
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