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