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Asterisk & VoIP

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

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  1. Asterisk & VoIP and it’s role in your enterprise

  2. 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

  3. 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

  4. 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

  5. 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

  6. Normal PBX Features • Name it, it’s there

  7. 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 (adam@bblisa.org) • 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, …

  8. 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

  9. 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 • …

  10. Possible Asterisk Configurations • Full PBX • Add small remote sites onto traditional PBX • Trunk Routing Gateway • Toll Bypass Only • Service Component

  11. Full PBX

  12. Traditional PBX with Remote Offices • Use of Digium appliance • Cisco x8xx routers

  13. Trunk Routing Gateway • Connect a traditional PBX to services like Sprint SIP Trunking

  14. Toll Bypass Only • Connect Traditional PBXs together over WAN Links

  15. Service Component • Meetme Conferencing Bridge • Voicemail / Unified Messaging • Add Softphones to non-VoIP PBX • …

  16. 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

  17. 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

  18. 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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