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    Download It Download It Presentation Transcript

    • Asterisk & VoIP and it’s role in your enterprise
    • Asterisk?
      • Open-source software released under the GPL
      • Sponsored by Digium, the main hardware provider for POTS interface cards
      • Digium named in the top 10 open source companies to watch by networkworld.com
      • Ports for most *nix systems including Solaris
      • Ports also available for OSX and Windows
      • Open standards along with some proprietary protocol support (like Cisco’s Skinny and MGCP)
      • Modular plugin type system
    • Protocols Supported
      • SIP – Session Initiation Protocol
      • H.323 – Common in video conferencing
      • Skinny – Cisco IP Phones default protocol
      • MGCP – Media Gateway Control Protocol
      • IAX - Inter-Asterisk Exchange Protocol
      Codecs Supported
      • G.711 – Best voice quality ~100Kbps
      • G.729 – Good voice quality ~40Kbps
      • GSM – Acceptable voice quality ~10Kbps
      • G.722, G.723.1, G.726, iLBC, Linear, LCP-10, Speex
    • PSTN Interface Support
      • Analog
        • FXO
        • FXS
        • E&M (w/ or w/o Wink), Loop start, Ground start, Kewl start
      • T1
        • E&M (w/ or w/o Wink)
        • Robbed bit
      • ISDN (PRI & BRI)
        • 4ESS
        • Lucent
        • National
        • Some international support
    • Biggest Features
      • Unified Voicemail
        • Voicemail to email (.wav files)
        • Video Conferencing messages to email (.mpg/.avi files)
      • Advanced Meetme conferencing - conference bridging
        • Web management interface
        • Support for conference numbers and passwords
        • Presenter and Presentee support (presenter can mute all participants, etc)
      • Contact Center Queuing
      • Interactive Voice Response
      • Automated Attendant
      • Video Conferencing (SIP and H.323)
      • Jabber / Google Talk integration
      • Find me / Follow me
      • Out of state DIDs (all VoIP systems)
      • Call monitoring and recording
    • Normal PBX Features
      • Name it, it’s there
    • Enterprise Features
      • Unified dialplan across many servers
        • SQL Compliant databases (through ODBC)
        • Native support for Mysql
        • LDAP integration
      • DNS SRV records
        • Make calling each other easy ( [email_address] )
        • Scalability
        • Load balancing
        • Phone provisioning
      • DUNDI – Distributed Universal Number Discovery
      • Works with SIP proxys to accept large amounts of phone registrations
      • AGI – Extend your system using C, C++, Perl, PHP, …
      • Custom CDR – Used for calling card integration, billing, …
    • Asterisk Compatible Endpoints
      • Cisco IP Phones (except IP 7920)
      • Polycom IP Phones
      • Snom IP Phones
      • Avaya IP Phones
      • Linksys IP Phones
      • Many others
      Asterisk Compatible Gateways
      • Cisco VoIP Gateways (anything MGCP, SIP, H.323)
      • Cisco Callmanager (through SIP and H.323)
      • Patton Smartnode
    • Asterisk PSTN Interfaces
      • Digium
        • Analog (up to 24 channels on a single PCI card, FXO and FXS) with hardware echo cancellation
        • T1 / PRI / BRI – 1-4 on a single PCI card with hardware echo cancellation
      • Sangoma
        • Better analog support, but uses more PCI slots (or spaces)
        • T1 / PRI / BRI – 1-8 on a single PCI card with hardware echo cancellation
        • Clear channel DS3
      • Any SIP / H.323 compliant endpoint
        • Cisco x8xx series ISR routers
    • Possible Asterisk Configurations
      • Full PBX
      • Add small remote sites onto traditional PBX
      • Trunk Routing Gateway
      • Toll Bypass Only
      • Service Component
    • Full PBX
    • Traditional PBX with Remote Offices
      • Use of Digium appliance
      • Cisco x8xx routers
    • Trunk Routing Gateway
      • Connect a traditional PBX to services like Sprint SIP Trunking
    • Toll Bypass Only
      • Connect Traditional PBXs together over WAN Links
    • Service Component
      • Meetme Conferencing Bridge
      • Voicemail / Unified Messaging
      • Add Softphones to non-VoIP PBX
    • Drawbacks
      • No PCI-Express Support (just came out for digital interfaces, still none for analog interfaces)
      • Hardware sizing information hard to find
      • Kernel updates break Digium drivers
      • PCI Bus sharing can cause significant problems with voice quality
      • NAT Traversal (common across any SIP system)
      • Linux system QoS not very mature yet
      • No VoIP security yet, although planned
      • Not for *nix beginners
    • Support
      • Certifications – dCAP (Digium Certified Asterisk Professional)
      • Voip-info.org (VoIP WIKI site)
      • Commercial support (through Digium)
      • Partners (certified or un-certified)
      • Mailing Lists (very active)
      • User Groups (although none in Boston)
      • Astricon (Asterisk conference)
      • Books
    • Where to start?
      • www.asterisk.org (Asterisk main page)
      • www.asterisknow.org (Asterisk and CentOS with full installer)
      • www.voip-info.org (huge resource for VoIP related projects and configuration info)
      • www.digium.com (hardware manufacturer and project sponsor)
      • Trixbox (formerly Asterisk @ HOME)
      • User Groups – Find one close or we can try to start one
      • Books – O’Reilly, Asterisk for Dummies, Asterisk configuration guide, …