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mr117
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mr115
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Admin manuals
Voice service configuration on PortaSIP (cloud PBX, SIP trunking, etc.)
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Cloud PBX (Hosted PBX) / Cloud Call Center solution
Cloud PBX concepts
Cloud Call Center solution overview
Additional authorization for toll calls
Auto attendant
Call barring
Call forking
Call forwarding
Call identification in Mobile PBX
Call parking
Call queues
Call recording
Call screening
Call supervision
Call transfer
Custom ringback tone
Voicemail
Instant messaging
Multiple pickup groups
Paging/Intercom calls
Hunt groups
PBX feature management
PBX feature summary
“Phone book” for each phone line
Shared line appearance
Stop/resume receiving hunt group calls
PortaSIP: SBC
PortaSIP: SBC
NAT traversal guidelines
NAT keepalive
Support of end points on IPv6 networks
Traffic density control
Interactive Connectivity Establishment (ICE) support
Secure calling
PortaSIP and emergency services (E911)
Route emergency calls passing user location data from MS Teams and other endpoints that support PIDF-LO
Legal call intercept
Support of STIR/SHAKEN standards
Service policies
Service policy configuration for ringback tone generation and early media relaying
PortaSIP as the IMS Telephony Application Server
SIP Trunking
Overview
Call delivery to a PBX with dynamic IP address
Delivering calls to customers with redirect servers
Simultaneous call limit for off-PBX calls
Authorization, charging and billing entities
Bandwidth-based simultaneous call limit
Cloud PBX call rating
Call authorization rules
Enabling product self-activation by a customer
Keepalive call monitoring
Limiting per-account simultaneous calls on a product
Special access codes
Special destinations
User authentication
Voice on-net rating
Fraud protection tools
Tools for prevention of VoIP fraud
Protection from DoS attacks
Toll fraud prevention
Fraud traffic profiles
Binding a user session to an IP address
Legacy services
Callback
Media server/IVR
Built-in IVR applications
Presence/BLF
Busy Lamp Field (BLF)
SIP devices with supported BLF
Presence
Voice calls processing
Voice calls processing
Call control via IVR
Call flow between multiple PortaSIP clusters
Call process/supported services
Calls from vendor via SIP
Number translation
Understanding SIP call routing
Video calls via SIP
Troubleshooting
Tracing unwelcome calls
Call quality monitoring
PortaSIP error codes
Service announcements via the media server
Active call monitoring
Advanced features
Integration with Mida Queue Manager
SIP identity
Support for privacy flags
SIP TAPI
Special prompt for calls to ported number
Caller ID (CNAM) lookup
“Diversion” removal on multiple call forwarding
API for computer-telephony integration services
Auto-provisioning IP phones
Successful SIP phone activation greeting
Bandwidth-aware limitation on concurrent calls permitted
Fax transmission-supported codecs
2nd number service
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