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کتابچه راهنمای VoIP: ساخت زیرساخت مخابراتی خودتان

VoIP Cookbook : Building your own Telecommunication Infrastructure

معرفی کتاب «کتابچه راهنمای VoIP: ساخت زیرساخت مخابراتی خودتان» (با عنوان لاتین VoIP Cookbook : Building your own Telecommunication Infrastructure) نوشتهٔ Onno W. Purbo, Anton Raharja، منتشرشده توسط نشر 2010 در سال 2010. این کتاب در فرمت pdf، زبان انگلیسی ارائه شده است.

ABOUT THE AUTHORS......Page 6 PREFACE......Page 7 How VoIP Works for Dummies.......Page 8 Where to Start?......Page 9 What Is Internet Telephony?......Page 10 PC to PC Internet Telephone Call......Page 12 Installing X-Lite......Page 16 X-lite Configuration......Page 20 Configuring Ekiga......Page 24 Configuring Account in Ekiga......Page 32 Test your SIP Softphone......Page 35 CHAPTER 3: VoIP Hardware for experienced Users......Page 40 Linksys PAP-2 Analog Telephone Adapter......Page 41 Linksys IP Phone SPA 941......Page 46 WiFi IPPhone......Page 51 Linksys Wireless-G IP Phone......Page 52 Activating Ipaq 6395's Wireless Capability......Page 61 Running SJPhone......Page 63 Using SJPhone to place call through Ipaq 6395......Page 70 Nokia......Page 73 Nokia Wireless Configuration......Page 74 SIP Server and Account Configuration in Nokia E61......Page 78 Internet Telephone Configuration in Nokia......Page 81 Registering to VoIP Softswitch......Page 82 Calling using Internet Telephone in Nokia E61......Page 85 VoIP in ADSL Modem......Page 87 ADSL Modem Configuration......Page 88 VoIP Configuration in Linksys WAG54GP2......Page 91 CHAPTER 4: Interconnectivity and Telephone Number Allocation......Page 98 Getting Free Washington State Telephone Number......Page 99 Free Internet Country: Country Code +882......Page 102 Introducing your country code to International VoIP network......Page 109 VoIP Rakyat's ENUM ......Page 111 Connecting to PSTN and Cellular Using VoIP Discount......Page 121 VoIP Cheap......Page 123 CHAPTER 5: Asterisk Softswitch......Page 125 Asterisk Installation......Page 126 ENUM.CONF Configuration......Page 127 EXTENSIONS.CONF Configuration......Page 128 Asterisk as SIP Client......Page 131 Generic SIP configuration......Page 133 CHAPTER 7: Briker Softswitch......Page 144 Zaptel Configuration......Page 155 SIP Trunk......Page 156 IAX2 Trunk......Page 159 H323 Trunk......Page 161 ZAP Trunk......Page 163 Outbound Routes......Page 164 Inbound Routes......Page 166 Setup Recordings......Page 167 Ring Groups......Page 168 Pin Sets......Page 170 Compile OpenSIPS......Page 171 Prepare User Database Server......Page 172 Use opensipsctl......Page 173 How to route to PSTN and Cellular......Page 174 How to route ENUM Query in OpenSIPS......Page 175 Test ENUM Query in OpenSIP......Page 176 ENUM Routing Table in OpenSIPS configuration......Page 177 Delegation Concept in ENUM......Page 179 Setup BIND for ENUM Server......Page 181 Test DNS for ENUM Query......Page 183 Configuring Conference Room MeetMe......Page 184 Configuring Dialplan for Conference......Page 185 Activating Conference while Operating......Page 186 CHAPTER 11: Trunk Peering in Asterisk......Page 188 CHAPTER 12: NAT and Firewall......Page 189 CHAPTER 13: Voicemail in Asterisk......Page 191 Attaching context......Page 194 The Extension Pattern......Page 195 Predefined Extension Names......Page 196 Defining Extension......Page 197 An interesting Extension Examples......Page 199 Forwarding to another Asterisk......Page 201 Linksys SPA9000......Page 203 Linksys SPA9000 Configuration......Page 204 Configuring VoIP on Linksys SPA9000......Page 207 CHAPTER 16: Analog Telephone Adapter for connection to PSTN......Page 212 Linksys SPA3000 Analog Telephone Adapter......Page 213 Configure Linksys SPA3000......Page 214 Linksys SPA3000 ATA Status......Page 218 LevelOne VOI-2100 Analog Telephone Adapter......Page 220 Using the SPA400 with Asterisk......Page 239 Configure Asterisk to talk to Linksys SPA400......Page 241 Connect PSTN using Linksys SPA9000 and Linksys SPA400......Page 244 Configure Linksys SPA9000 to talk to Linksys SPA400......Page 253 CHAPTER 17: Peering Among Providers......Page 254 Becoming a Peer in SIP Network ......Page 256 Coding Decoding (CODEC)......Page 258 Mean Opinion Score (MOS)......Page 259 MOS and R Factor values for G.711, G.723, and G.729......Page 261 Calculating The Required Bandwidth......Page 262 Calculation for Call Center......Page 265 VQManager Installation......Page 271 Some of the Important Scripts of VQManager......Page 272 Activate VQManager Web Service......Page 273 Inserting new Interface......Page 281 Monitor VoIP Performance......Page 282 Installation of SIPp Webfrontend......Page 291 Transaction Oriented Test using SIPp......Page 292 Access to the SIPp Webfrontend......Page 295 CODEC and Vocoder......Page 306 Test prior to operation of the system......Page 307 Some Useful References For VoIP Troubleshooting......Page 308 Testing Software......Page 309 APPENDIX A: Example of /etc/sip.conf......Page 310 APPENDIX B: SIPp COMMANDS......Page 320 APPENDIX C: File /usr/local/etc/opensips/cfg-test-uas.cfg......Page 327
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