Switching to VoIP

Switching to VoIP

by Theodore Wallingford
Released June 2005
Publisher(s): O'Reilly Media, Inc.
ISBN: 9780596517298

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Book description

More and more businesses today have their receive phone service through Internet instead of local phone company lines. Many businesses are also using their internal local and wide-area network infrastructure to replace legacy enterprise telephone networks. This migration to a single network carrying voice and data is called convergence, and it's revolutionizing the world of telecommunications by slashing costs and empowering users. The technology of families driving this convergence is called VoIP, or Voice over IP.

VoIP has advanced Internet-based telephony to a viable solution, piquing the interest of companies small and large. The primary reason for migrating to VoIP is cost, as it equalizes the costs of long distance calls, local calls, and e-mails to fractions of a penny per use. But the real enterprise turn-on is how VoIP empowersbusinesses to mold and customize telecom and datacom solutions using a single, cohesive networking platform. These business drivers are so compelling that legacy telephony is going the way of the dinosaur, yielding to Voice over IP as the dominant enterprise communications paradigm.

Developed from real-world experience by a senior developer, O'Reilly's Switching to VoIP provides solutions for the most common VoIP migration challenges. So if you're a network professional who is migrating from a traditional telephony system to a modern, feature-rich network, this book is a must-have. You'lldiscover the strengths and weaknesses of circuit-switched and packet-switched networks, how VoIP systems impact network infrastructure, as well as solutions for common challenges involved with IP voice migrations. Among the challenges discussed and projects presented:

  • building a softPBX
  • configuring IP phones
  • ensuring quality of service
  • scalability
  • standards-compliance
  • topological considerations
  • coordinating a complete system ?switchover?
  • migrating applications like voicemail and directoryservices
  • retro-interfacing to traditional telephony
  • supporting mobile users
  • security and survivability
  • dealing with the challenges of NAT

To help you grasp the core principles at work, Switching to VoIP uses a combination of strategy and hands-on "how-to" that introduce VoIP routers and media gateways, various makes of IP telephone equipment, legacy analog phones, IPTables and Linux firewalls, and the Asterisk open source PBX software by Digium.You'll learn how to build an IP-based or legacy-compatible phone system and voicemail system complete with e-mail integration while becoming familiar with VoIP protocols and devices. Switching to VoIP remains vendor-neutral and advocates standards, not brands. Some of the standards explored include:

  • SIP
  • H.323, SCCP, and IAX
  • Voice codecs
  • 802.3af
  • Type of Service, IP precedence, DiffServ, and RSVP
  • 802.1a/b/g WLAN

If VoIP has your attention, like so many others, then Switching to VoIP will help you build your own system, install it, and begin making calls. It's the only thing left between you and a modern telecom network.

Table of contents

  1. Table of Contents (1/2)
  2. Table of Contents (2/2)
  3. Foreword
  4. Preface
    1. Audience
      1. Regarding Asterisk
    2. Assumptions Made in This Book
    3. Conventions
    4. Where to Get More
    5. Safari Enabled
    6. How to Contact Us
    7. Acknowledgments
  5. Voice and Data: Two Separate Worlds?
    1. The PSTN
      1. Mesh Versus Switched
      2. Signaling System 7
      3. Plain Old Telephone Service
    2. Key Systems and PBXs
      1. Lines and Trunks
    3. Limits of Traditional Telephony
    4. VoIP in the Home
    5. VoIP in Business
    6. VoIP’s Changing Reputation
    7. Key Issues: Voice and Data: Two Separate Worlds
  6. Voice over Data: Many Conversations, One Network
    1. VoIP or IP Telephony
      1. VoIP’s Pros and Cons
      2. VoIP Network Fundamentals
      3. The Layers of a VoIP Network
        1. The physical layer
        2. The data link layer
        3. The network layer
        4. The transport layer
        5. To connect or not to connect
        6. The session, presentation, and application layers
      4. VoIP Servers
      5. Voice Endpoints
        1. IP phones or traditional phones
      6. Project 2.1. Configure an IP Hardphone and the VoIP Test Network
        1. Configuring a Grandstream Budgetone 101 IP phone
        2. A simple VoIP test network
      7. Project 2.2. Make an IP-to-IP Phone Call
    2. Distributed Versus Mainframe
      1. The Core and the Edge
      2. VoIP in Enterprise Networks
      3. Network Convergence
      4. Pure IP or IP Enabled
    3. Key Issues: Voice over Data: Many Conversations, One Network
  7. Linux as a PBX
    1. Free Telephony Software
      1. Other Free Telephony Software
      2. Asterisk’s Requirements
    2. Installing Legacy Interface Cards
      1. The X100P Card
        1. The POTS pass-through connector
        2. Installing an X100P
      2. The TDM400P Card
      3. Other Interfaces
    3. Compiling and Installing Asterisk
      1. Install the Software Components of Asterisk
      2. For Those Who Prefer RPM Packages
      3. Loading the Interfacing Drivers
      4. Starting and Stopping the Asterisk Server
      5. Configuring for Automatic Startup at Boot Time
        1. Using rc.local to launch Asterisk
        2. Using the safe_asterisk script
        3. Setting up a Red Hat init script for Asterisk
      6. Securing the Asterisk Instance
      7. Asterisk on Mac OS X
      8. Project 3.1. Test an IP Phone with Asterisk
        1. Verify Asterisk’s SIP server is running
        2. Set the IP phone to use a SIP server
        3. Allow the IP phone to place calls via Asterisk
        4. Restart or reload?
        5. Listen to the Asterisk automated greeting using an IP phone
        6. Listen to a voice over Internet greeting
      9. Installing Mpg123
    4. Monitoring Asterisk
      1. Asterisk’s logfiles
      2. Astman
      3. Channels
    5. Key Issues: Linux as a PBX
  8. Circuit-Switched Telephony
    1. Regulation and Organization of the PSTN
      1. The FCC
      2. The International Telecommunications Union
      3. RBOCs and CLECs
        1. LD carriers
        2. IXCs
        3. ISPs
    2. Components of the PSTN
      1. The Central Office
      2. PSTN Distribution Frames
      3. Main Distribution Frames
      4. Switch-to-Switch Trunks
      5. In-Band Signaling (DTMF)
        1. Standard tones
        2. City codes, area codes, and country codes
      6. Out-of-Band Signaling and SS7
      7. PRI/T1
      8. SONET and DS3
      9. DID
      10. Hunt Groups
      11. Centrex
    3. Customer Premises Equipment
      1. Switches
      2. The Demarc
      3. “Inside Wiring” and the Cable Plant
      4. CP Distribution Frames
      5. FXS and FXO Signaling
      6. Loop Start Signaling
      7. Channel-Associated Signaling
      8. Other Legacy Standards
      9. Project 4.1. Create a Trunk Channel with a POTS Line
        1. What you need for this project:
    4. Time Division Multiplexing
      1. Pulse Code Modulation and DS0 Channels
      2. Channel Banks
      3. BRI
    5. Point-to-Point Trunking
      1. Copper Cabling
      2. Fiber-Optic Cabling
      3. Radio
      4. Free-Space Optics
    6. Legacy Endpoints
      1. FXS/Analog Endpoints
      2. Digital Endpoints
        1. Using (or not using) digital endpoints with a VoIP server
        2. Speakerphone
        3. Lines and line appearances
        4. Ring groups and private hunt groups
      3. Project 4.2. Simulate a Simple Key System with a Ring Group
        1. What you need for this project:
        2. Step 1: Unlock and configure phone A (Cisco 7960)
        3. Step 2: Configure phone B (Budgetone 101)
        4. Step 3: Define SIP peers for the IP phones
        5. Step 4: Create a ring group for the two IP phones
        6. Step 5: Allow the SIP phones to dial out
    7. Dial-Plan and PBX Design
      1. Extension Numbering
        1. Extensions based on DID
        2. Extensions based on geography
        3. Extensions based on department
        4. Extensions based on type of device
        5. Extensions “just because”
      2. Dial-Tone Trunks
    8. Key Issues: Circuit-Switched Telephony
  9. Enterprise Telephony Applications
    1. Application Terminology
    2. Basic Call Handling
      1. Intercom Call
      2. Mute, Hold, Call Transfer, and Multiparty Conference
        1. Blind and consultative transfer
        2. Conference
        3. Meet-me
      3. Caller ID
    3. Administrative Applications
      1. Call Accounting
      2. Project 5.1. Analyze Call Detail Records and Call Accounting
        1. What you need for this project:
        2. Asterisk CDR default fields
      3. Console
      4. In-Out, DND, and Call Forward
      5. Call Logs and Missed-Call Indications
    4. Messaging Applications
      1. Overhead Paging
      2. Barging
      3. Voice Mail
        1. Message notification (pager, email, etc.)
    5. Advanced Call-Handling Applications
      1. Call Parking and Orbit
      2. Project 5.2. Set Up Call Parking
        1. Put that call in orbit
      3. Automatic Call Return
      4. Hunt Groups and Ring Groups
      5. Project 5.3. Set Up a Private Hunt Group
        1. What you need for this project:
        2. Add cell phone bridging to this hunt group
      6. Hold Queues
      7. Directories
      8. Presence
      9. Bridging
    6. CTI Applications
      1. Automated Attendants
        1. Interactive voice response and data collection
        2. Privacy management
      2. Call Centers
    7. Key Issues: Telephony Applications
  10. Replacing the Voice Circuit with VoIP
    1. The “Dumb” Transport
    2. Voice Channels
      1. Sampling and Digitizing
        1. The 64 kbps channel
      2. Encoding (1/2)
      3. Encoding (2/2)
        1. Framing
        2. Digital versus packet based
        3. Multiplexing
        4. Compression
        5. Codecs
        6. Codec packet rates
      4. Transport
        1. The T1 carrier versus VoIP
        2. Voice packet structure
        3. Real-Time Transport Protocol
        4. Ethernet
      5. Decoding and Playback
        1. Things that degrade playback quality
        2. Transcoding
        3. Call paths
        4. Silence suppression and comfort noise generation
      6. Project 6.1. Set Up Custom Codec Selection and Enable an Independent Call Path
        1. What you need for this project:
        2. Per-peer codec selection on Cisco media gateways
        3. Per-peer codec selection on Asterisk
    3. Key Issues: Replacing the Voice Circuit with VoIP
  11. Replacing Call Signaling with VoIP
    1. VoIP Signaling Protocols
      1. H.323, SIP… How Do I Choose?
        1. Signaling for non-telephony apps
    2. H.323
      1. H.323 Gatekeeper
        1. Registration
      2. H.323 Terminal
      3. H.323 Gateway
      4. Multipoint Control Units
      5. The H.323 Call-Signaling Process
        1. Setup/teardown
        2. Capabilities negotiation
        3. Open media channel
        4. Perform call
        5. Release
      6. E.164 Address Scheme
      7. H.245
        1. Fast-start
        2. SoftPBX H.323 implementations
      8. Project 7.1. Build an H.323 Gatekeeper Using Open H.323 (1/2)
      9. Project 7.1. Build an H.323 Gatekeeper Using Open H.323 (2/2)
        1. What you need for this project:
        2. Open H.323’s requirements
        3. Download and compile Open H.323
        4. Set up the Gnu gatekeeper (gnugk)
        5. Register an H.323 softphone using OhPhoneX
        6. Register an H.323 endpoint using NetMeeting
        7. Make the call
    3. SIP
      1. SIP Nodes (1/2)
      2. SIP Nodes (2/2)
        1. SIP registrar
        2. URIs
        3. SIP methods and responses
        4. SIP proxies
        5. SIP user agent elements
        6. SIP redirect
        7. Session Description Protocol
        8. Real-Time Streaming Protocol
        9. SIP packet encoding
        10. SIP versus H.323: the great debate
    4. IAX
    5. MGCP
      1. MEGACO/H.248
    6. Cisco SCCP
    7. Heterogeneous Signaling
      1. Heterogeneous Signaling and the SoftPBX
        1. Cisco AVVID/CallManager
        2. Asterisk
    8. Key Issues: Replacing Call Signaling with VoIP
  12. VoIP Readiness
    1. Assessing VoIP Readiness
    2. Business Environment
      1. The Market Adoption Cycle
      2. Efficiency
      3. Productivity
      4. Cost
        1. Cost models help sell IP telephony
        2. Actual consumption cost model
        3. Overhead costs
        4. The success delta
        5. Service provider cost savings
        6. Internal management cost savings
        7. How do you eat an elephant?
        8. Recognizing revenue, productivity, and cost reduction
        9. Calculate the ROI
      5. Convenience and Timing
    3. Network Environment
      1. LAN
        1. VoIPX? VoNetBEUI? Not a chance.
        2. Cabling
        3. Switches versus repeaters
        4. Wireless Ethernet
      2. Power over Ethernet
        1. Cisco power versus 802.3af
        2. Power injection
      3. LAN Readiness Checklist
      4. WAN
      5. VPN
        1. Managed VPN
        2. VoIP over dial-up
    4. Implementation Plan
      1. Knowledge Foundation
      2. Time
      3. Manpower, Vendors, and VARs
        1. Selecting a VoIP platform
        2. Choosing a VAR
        3. Creating an RFP
      4. Training
    5. Key Issues: VoIP Readiness
  13. Quality of Service
    1. QoS Past and Present
      1. Call-quality scoring
      2. How you can use the MOS scale
      3. Noise
      4. QoS is Two Things
        1. Does overcapacity remove the need for QoS?
    2. Latency, Packet Loss, and Jitter
      1. Echo
        1. Hybrids and echo
        2. Zaptel’s echo suppression
    3. CoS
      1. 802.1p and ToS
      2. DiffServ
        1. Policy servers
        2. DSCP classes
        3. The DiffServ CoS process
      3. Project 9.1. Create a DiffServ Decision Point with Linux
        1. What you need for this project:
        2. Configure an iptables edge router for DiffServ
        3. LTC for DiffServ core routers
        4. LTC on a softPBX
    4. 802.1q VLAN
      1. Layer 2 Switching
      2. Layer 3 Switching
        1. VLAN trunking
        2. CoS over VLAN trunks
    5. Quality of Service
      1. Intserv and RSVP
        1. Controlled loads versus guarantees
      2. MPLS
    6. Residential QoS
      1. Dial-Tone Providers That Offer VoIP Service
      2. Residential and Small-Office VoIP Routers
    7. Voice QoS on Windows
      1. QoS Packet Scheduler
      2. Windows RSVP Service
      3. Project 9.2. Audit a Network’s Capabilities Using pathping and traceroute
        1. What you need for this project:
        2. Measure the latency time and jitter on a call path
    8. Best Practices for Quality of Service
    9. Key Issues: Quality of Service
  14. Security and Monitoring
    1. Security in Traditional Telephony
      1. Access Control
        1. Snooping
        2. Phreaking
      2. Call Accounting and Billing
      3. Features
    2. Security for IP Telephony
    3. Access Control
      1. Credentials and Authentication
      2. Project 10.1. Use MD5 Hash to Secure SIP Passwords
        1. What you need for this project:
        2. Origin, destination, and timing policies
        3. Higher-layer access control
        4. Physical presence policies
      3. Media Encryption
    4. Software Maintenance and Hardening
      1. Hardening VoIP Servers
      2. Project 10.2. Harden a SoftPBX
        1. What you need for this project:
        2. Remove unnecessary software
        3. Clean up xinetd
        4. Kill shell access for daemon users
        5. Optimize the local firewall on the softPBX
        6. Check for security risks in the dial-plan
      3. DMZs and Firewalls
        1. Building a DMZ
    5. Intrusion Prevention and Monitoring
      1. Project 10.3. Logging and Controlling VoIP Packets with iptables
        1. What you need for this project:
        2. Reading and analyzing packet logs
      2. SNMP
      3. Project 10.4. Tune Up Asterisk’s Logging Configuration
        1. What you need for this project:
        2. The console keyword
        3. Use a non-default log directory
        4. Enable syslog
      4. Snort and Nagios
    6. Key Issues: Security and Monitoring
  15. Troubleshooting Tools
    1. VoIP Troubleshooting Tools
    2. The Three Things You’ll Troubleshoot
    3. SIP Packet Inspection
      1. Project 11.1. Inspect SIP Traffic with Ethereal
        1. What you need for this project:
        2. Configure the SIP softphone
        3. Configure Ethereal
        4. Observe SIP registration
        5. Observe registration failure
      2. Project 11.2. Inspect SDP Capabilities Negotiation
        1. What you need for this project:
        2. Inspect successful capabilities negotiation
        3. Inspect failed capabilities negotiation
    4. Interoperability
      1. Project 11.3. Trace Both Ends of a Call Setup with Log Comparison
        1. What you need for this project:
      2. Troubleshooting Quality-of-Service Issues
        1. Commercial packet analysis tools
        2. When the going gets tough, call in the end users
    5. When, Not if, You Have Problems…
    6. Simulating Media Loads
    7. Key Issues: Troubleshooting Tools
  16. PSTN Trunks
    1. Dial-Tone Trunks
      1. POTS and Centrex Trunks
        1. T1/PRI trunks
        2. ISDN BRI trunks
        3. VoIP trunks
        4. Hosted PBX
        5. ATM trunks
        6. Television cable and fiber trunks
        7. If that’s not enough bandwidth…
      2. How Many Dial-Tone Trunks Are Needed?
        1. The case for fewer trunks
        2. The case for more trunks
      3. Project 12.1. Make It Easier for Callers to Reach PBX Users
        1. What you need for this project:
        2. Home phone call bridging
        3. Time-based context includes
      4. More on Trunk Sizing
      5. Connecting Trunks to Your Telephone Network
      6. Channelized or Split-Use T1s
      7. Using the PSTN for Intraorganization Calls
        1. Leverage Centrex groups
        2. Use dial-tone trunks to seamlessly route calls between PBXs in the same organization
      8. Project 12.2. Use PSTN Trunks in a Multioffice Dial-Plan
        1. What you need for this project:
        2. Controlling Caller ID when using PSTN trunks
    2. Routing PSTN Calls at Connect Points
      1. Project 12.3. Grouping PSTN Trunks
        1. What you need for this project:
      2. Project 12.4. Create an Automatic Call Distribution (ACD) Scheme Based on Area Code
        1. What you need for this project:
        2. What if the call is from a different area code?
        3. What if the caller is using caller ID blocking?
      3. Project 12.5. Use Distinctive Ring Detection
        1. What you need for this project:
        2. Routing based on distinctive ring
    3. Timing Trunk Transitions
    4. Key Issues: PSTN Trunks
  17. Network Infrastructure for VoIP
    1. Legacy Trunks
      1. Private Analog Lines
        1. Leased lines
        2. Dry lines
      2. Private Digital Trunks
    2. VoIP Trunks
      1. Unwanted Effects of Load Management
        1. Traffic diversion (failover circuits)
        2. Load-splitting
        3. Multipath jitter
        4. Multilink PPP
      2. TCP/IP as a Transport for Voice Trunks
        1. Insecure, unencrypted UDP
        2. VPN
        3. GRE tunnels
        4. IP addressing schemes
      3. Supporting VoIP Road Warriors
        1. To VPN or not to VPN?
        2. Soft- or hardphone in the mobile office?
      4. Project 13.1 Use Dial-Plan to Connect to Multiple VoIP Networks
        1. What you need for this project:
        2. The Inter-Asterisk Exchange Telephone (IAXTel) network
        3. Get access to IAXTel
        4. Set up the dial-plan for regular PSTN calls
        5. Route toll-free calls to the Internet
        6. Allow incoming calls from IAXTel
        7. Route incoming IAXTel calls
        8. Monitoring registrations
    3. WAN Design
      1. Distributed Versus Client/Server
      2. WAN Layout
        1. Hub and spoke
        2. Peered
        3. Meshed
      3. Layout and PBX Placement
        1. Locate to conserve network availability
        2. Locate to save money
        3. Locate for capabilities
        4. Don’t locate for convenience
    4. Disaster Survivability
      1. Surviving Power Failures
        1. Multiphase power
        2. Uninterruptible power supplies
      2. Surviving Network Link Failures
        1. PSTN trunk failures
      3. Remote Site Survivability and ALS
        1. The survivability problem
        2. The survivability solution
    5. Metro-Area Links
    6. Firewall Issues
      1. DMZ Eliminates the Need for NAT
      2. IAX Eliminates the Need for NAT
      3. STUN Allows Coexistence with NAT
      4. VPN Allows Coexistence with NAT
    7. Peer-by-Peer Codec Selection
      1. Controlling Codec Selection in Asterisk SIP Peers
      2. Directory Services
        1. Directory services: resolution and advertising
        2. Resolution of SIP URIs and E.164 numbers
        3. ENUM
        4. DUNDi
        5. Advertising
      3. Project 13.2. Build an Interactive Directory on a SIP Display Phone
        1. What you need for this project:
    8. Key Issues: Network Infrastructure for VoIP
  18. Traditional Apps on the Converged Network
    1. Fax and Modems
      1. T.30, T.37, and T.38
        1. ATAs and fax/modem support
      2. Project 14.1. Turn Your Linux Box into a Fax Machine
        1. What you need for this project:
        2. Receiving faxes
        3. Sending faxes
      3. Project 14.2. Build an Inbound Fax-to-Email Gateway
        1. What you need for this project:
        2. Automatic fax routing
        3. Fax routing with DIDs
        4. For those who prefer PDF files
        5. Simulating T.37
      4. FoIP Solutions
        1. Cisco AVVID
        2. Avaya Media Servers
        3. Other solutions
        4. Commercial soft fax solutions
        5. Hylafax
      5. Hosted Fax
    2. Fire and Burglary Systems
    3. Surveillance Systems and Videoconferencing
      1. Camera Surveillance
      2. Videoconferencing
    4. Voice Mail and IVR
      1. Project 14.3. Build a Custom Voice Mail System
        1. What you need for this project:
        2. Turn on email notification
        3. Attach voice mail sound files to email notifications
        4. Customize the email notification message
        5. Connect the dial-plan to voice mail
        6. Web access to voice mail
      2. Project 14.4. Create Custom Sounds to Interact with Callers
        1. What you need for this project:
        2. Use a prerecorded sound for a simple paging routine
        3. Use your own sounds in IVR
        4. Converting sounds using SoX
        5. Use soxmix to put background music into an announcement
        6. Record IVR announcements using Asterisk instead of a sound recording app
    5. Emergency Dispatch/911
      1. E911 on the PSTN
        1. E911 on wireless providers’ networks
        2. E911 on the converged network
        3. 911 on large VoIP networks
        4. Private Switch ALI
        5. Mapped ALI
        6. POTS pass-through
        7. Dial-around
      2. Project 14.5. Use VoIP Dial-Around to Connect 911 Calls
        1. What you need for this project:
      3. Administrator Tools
        1. Asterisk’s administrative interface
        2. AMP
        3. AM
        4. AstConsole and Asterisk setup assistants
        5. Open H.323 administrative GUIs
        6. Other administrator GUI tools
    6. Key Issues: Traditional Apps on the Converged Network
  19. What Can Go Wrong?
    1. Common Problem Situations
      1. The people you call complain about echo
      2. The phone rings, but callers cannot hear you
      3. SIP registrations don’t work through a firewall
      4. The IP phone can’t make any calls
      5. Past a certain number of simultaneous calls, quality breaks down or calls are disconnected
      6. You lose the dial-tone every few days or so, and you can’t receive any calls from the PSTN
      7. Dialed digits work to place calls but not to interact with IVR prompts
      8. Callers sound robotic, or they say you do
      9. Calls across a wide area call path have dropouts in the audio
      10. You and your caller find yourself interrupting each other a lot
      11. When a caller begins to speak, you can’t hear the first word or two of his sentences
      12. The power went out and so did all the phones
      13. I love the Cisco IP phones but they don’t accept 802.3af inline power. What do I do?
      14. When my PBX routes a call to IAXTel or another Internet voice destination, the sound quality is a...
      15. A clumsy keystroke took the softPBX down during peak business hours
      16. The old-timers are complaining about the new phones, the new voice mail greeting, or the new ____...
      17. My IP telephony salesperson said I would be able to do overhead or zone paging using the IP phone...
      18. The VoIP budget was, well, too small
      19. The phone company missed a critical circuit switchover deadline
      20. I’ve read all about QoS and proper converged network design, but I’m still paranoid about quality...
    2. Key Issues: What Can Go Wrong?
  20. VoIP Vendors and Services
    1. Softphones and Instant Messaging Software
      1. OhPhone
      2. X-Lite, X-Pro, and Eyebeam
      3. LIPZ4
      4. Firefly (SIP and IAX)
      5. IAXComm
      6. NetMeeting
      7. MSN Messenger
      8. iChat
    2. Skype
      1. Skype Web Links
      2. Conference Calls
      3. Skype Handset
      4. Other Softphones to Check Out
    3. Other Desktop Telephony Software
      1. Call Soft Pro
      2. CallerID
      3. GeckoPhone
      4. FaxMachine
      5. FaxStatus (Mac)
      6. PhoneValet (Mac)
      7. Voicent Gateway
      8. SpeechSoft
    4. Developer Tools and SoftPBX Systems
      1. Asterisk
        1. AstLinux
        2. Asterisk Management Portal
        3. LDAPGet
        4. TAFM
        5. Asterisk Perl modules
        6. JAsterisk
        7. AstWind
      2. Non-Asterisk
        1. Open H.323
        2. VOCAL
        3. Intel Dialogic products
        4. SIP Express Router
        5. Envox
        6. Bayonne
        7. VOCP
        8. Telos ISDN H.323 Gateway
      3. TFTP Servers and Tools
    5. VoIP Service Providers
      1. Dial-Tone and Call Origination
        1. Bulk trunk providers
        2. Wholesale IXCs
        3. Hosted PBX
        4. Web-based call center services
    6. Telephony Hardware Vendors
  21. Asterisk Reference
    1. How Asterisk Is Supported
    2. Asterisk’s Configuration Files
      1. Including Configuration Files
    3. Asterisk Dial-Plan
      1. Variables (1/2)
      2. Variables (2/2)
        1. Built-in variables
        2. The layout of extensions.conf
        3. Contexts
        4. Including contexts
        5. The syntax of extensions
        6. Extension pattern matching
        7. Special extensions
      3. String Processing in the Dial-Plan
        1. Getting string lengths
        2. Concatenating strings
      4. Dial-Plan Command Reference (1/4)
      5. Dial-Plan Command Reference (2/4)
      6. Dial-Plan Command Reference (3/4)
      7. Dial-Plan Command Reference (4/4)
        1. AbsoluteTimeout
        2. AddQueueMember
        3. ADSIProg
        4. AGI
        5. Answer
        6. AppendCDRUserField
        7. BackGround
        8. BackgroundDetect
        9. Busy
        10. ChangeMonitor
        11. ChanIsAvail
        12. CheckGroup
        13. Congestion
        14. ControlPlayback
        15. Cut
        16. DBdel
        17. DBdeltree
        18. DBget
        19. DBput
        20. DeadAGI
        21. Dial
        22. DigitTimeout
        23. Directory
        24. DISA
        25. Echo
        26. EnumLookup
        27. Festival
        28. Flash
        29. GetGroupCount
        30. Goto
        31. GotoIf
        32. GotoIfTime
        33. Hangup
        34. LookupBlacklist
        35. MailboxExist
        36. Math
        37. MeetMe
        38. MeetMeAdmin
        39. MeetMeCount
        40. Milliwatt
        41. Monitor
        42. MP3Player
        43. MusicOnHold
        44. NoCDR
        45. NoOp
        46. ParkedCall
        47. Playback
        48. Playtones
        49. PrivacyManager
        50. Queue
        51. Random
        52. Read
        53. ResetCDR
        54. ResponseTimeout
        55. Ringing
        56. Rpt
        57. SayAlpha
        58. SayDigits
        59. SayNumber
        60. SayPhonetic
        61. SayUnixTime
        62. SendDTMF
        63. SendText
        64. SendURL
        65. SetAccount
        66. SetCallerID
        67. SetCDRUserField
        68. SetCIDName
        69. SetCIDNum
        70. SetGlobalVar
        71. SetLanguage
        72. SetMusicOnHold
        73. SetVar
        74. SIPdtmfMode
        75. SMS
        76. SoftHangup
        77. StopMonitor
        78. StopPlaytones
        79. StripLSD
        80. StripMSD
        81. System
        82. Transfer
        83. VoiceMail
        84. VoiceMailMain
        85. Wait
        86. WaitExten
        87. WaitMusicOnHold
        88. Zapateller
        89. ZapBarge
        90. ZapScan
    4. Asterisk Channels
      1. Zaptel Channels
        1. Using analog interfaces with Asterisk
        2. Using T1 interfaces with Asterisk
      2. SIP Channels
        1. SIP peer configuration
    5. The Asterisk CLI
      1. Inspecting Channels
      2. Inspecting the Dial-Plan
      3. Administering the Dial-Plan Using the CLI
      4. CLI Command Reference (1/2)
      5. CLI Command Reference (2/2)
        1. Administrator commands
        2. Process control commands
        3. AGI commands
        4. Database commands
        5. IAX v2 commands
        6. IAX v1 commands
        7. Crypto commands
        8. H.323 commands
        9. SIP commands
        10. MGCP channel commands
        11. SCCP channel commands
        12. CAPI channel commands
        13. Zaptel channel commands
    6. Integrating Asterisk with Other Software
      1. Asterisk CLI Wrapper
      2. The Asterisk Gateway Interface
      3. Asterisk Manager Socket API
    7. Key Issues: Asterisk Reference
  22. SIP Methods and Responses
  23. AGI Commands
  24. Asterisk Manager Socket API Syntax
  25. Glossary (1/2)
  26. Glossary (2/2)
  27. Index (1/6)
  28. Index (2/6)
  29. Index (3/6)
  30. Index (4/6)
  31. Index (5/6)
  32. Index (6/6)

Product information

  • Title: Switching to VoIP
  • Author(s): Theodore Wallingford
  • Release date: June 2005
  • Publisher(s): O'Reilly Media, Inc.
  • ISBN: 9780596517298