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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

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asterisk voip

Asterisk & VoIP

and it’s role in your enterprise

asterisk
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
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
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
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
Normal PBX Features
  • Name it, it’s there
enterprise features
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 protected])
    • 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
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
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
Possible Asterisk Configurations
  • Full PBX
  • Add small remote sites onto traditional PBX
  • Trunk Routing Gateway
  • Toll Bypass Only
  • Service Component
traditional pbx with remote offices
Traditional PBX with Remote Offices
  • Use of Digium appliance
  • Cisco x8xx routers
trunk routing gateway
Trunk Routing Gateway
  • Connect a traditional PBX to services like Sprint SIP Trunking
toll bypass only
Toll Bypass Only
  • Connect Traditional PBXs together over WAN Links
service component
Service Component
  • Meetme Conferencing Bridge
  • Voicemail / Unified Messaging
  • Add Softphones to non-VoIP PBX
drawbacks
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
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
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, …
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