Leverage Existing PBX Infrastructure with Asterisk

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Leverage Existing PBX Infrastructure with Asterisk
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The VoIP Migration Problem

• 365 Million Digital PBX Handsets
• Handsets, LAN, Wiring, Knowledge Base
• US100 Billion Existing PBX Infrastructure

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Enterprise VoIP Landscape

• VoIP Not If Question, Rather a When Question
• Digium/Asterisk Open Architecture Maximum
  ROI/Maximum Customization
• Now About That Existing PBX Investment...

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Enterprise PBX Landscape

• Hundreds of Digital PBX Handset Models
• Each Has Unique PBX-to-Handset Signaling
• Not Open, Interoperable Between, Among Brands

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Whats In Common?

• Most cases, use Cat 3 wiring
• Users are well trained comfortable
• Devices do their job voice

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Advent of IP Telephony

• Voice Becomes Application on VoIP Platform
• Software-based Infrastructure Much More Flexible
  than Hardware-based Infrastructure
• Rich Applications, Cost-savings, Productivity

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The Conduit SIP

• SIP Enables Common, Open Interface
• Provides Multi-Vendor Integration Across
  Platforms, Distributed Locations
• Converged, Next Gen Apps on the Laptop/ Desktop,
  Voice Stays Where it Belongs

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Asterisk Digital PBX Handsets

• In Phone Closet PBX Hardware Removed
• Digital PBX Handsets remain in place over Cat 3
  wiring to patch panels
• Amphenol from patch panels to SIP Handset Gateway
  to Asterisk server platform, whether premise or
  hosted

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Asterisk Digital PBX Handsets

• Digital PBX Handsets Become SIP Handsets with
  Asterisk Call Control

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SIP Specifications Supported
RFC 2617 - HTTP Authentication Basic and Digest
Access Authentication
RFC 3261 - Session Initiation Protocol
RFC 2705 Media Gateway Control Protocol (used
for Digit Map implementation)
RFC 2833 - RTP Payload for DTMF Digits
RFC 3265 - SIP-Specific Event Notification
draft-ietf-sip-refer-07-Refer-To Header
RFC 1321 - MD5 Message-Digest Algorithm
RFC 3264 - An Offer/Answer Model with SDP
RFC 783 - TFTP Protocol (used for transferal of
configuration files to the gateway)
draft-ietf-sipping-mwi-01 - Message Waiting
Indication
draft-burger-sipping-netann-05
draft-ietf-sipping-cc-transfer-01
RFC 2327 - SDP Session Description Protocol
draft-ietf-sipping-dialog-package-01
draft-ietf-sipping-service-examples-04
RFC 1889 - RTP Transport Protocol for Real-Time
Applications
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Case Study GiftTree.com

• Operates Network of 200 Floral, Gift Websites.
  60K unique orders Holiday 2005
• LD, Toll Cost Savings Achieved. No integration
  between Nortel PBX, Web, Oracle Database
• 40 Call Center Employees, More in Peak Months

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Case Study GiftTree.com

• Bid for call center IP PBX upgrade US40K.
  However, concern over cutover, re-training
• Decision made to deploy Asterisk IP PBX, 2 Citel
  SIP Handset Gateways
• Total Equipment Cost for Asterisk, Citel ltUS10K

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Cutover Time?
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Five Minutes.
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Enterprise Benefits Cost

• 70 Cost Savings over Rip Replace
• Additional Users Added with Minimal Investment
• Minimal Business Disruption

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Enterprise Benefits Productivity

• Web, Database, Voice Applications Integrated
• New VoIP Functionality on Existing PBX Phones
• Little or No Retraining

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Summary

• Enterprise VoIP Migration Closer Than You Think
• Asterisk Flexibility, Citel Leverage
• Business Case Unique for Every Enterprise

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Questions, Comments?
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Thank you.Visit citel.com/10myths