Our Internet Telephony Service Provider (ITSP) will send the calls incoming from DID numbers to us on our "external" SIP profile (port 5080 in demo configuration) .
We will send to the ITSP outbound calls in need of a SIP routing to the external world, using the "external" SIP profile. In dialplan we will use a SIP "gateway" syntax that refers to the authentication needed by our ITSP to accept our outbound traffic.
- Preface
- Architecture of FreeSWITCH
- Building and Installation
- Test Driving the Example Configuration
- Important concepts to understand
- FS_CLI and Console, Controlling FreeSWITCH
- Configure SIP and make the first calls in demo dialplan
- Summary
- User Directory, SIP, and Verto
- WebRTC, SIP, and Verto
- XML Dialplan
- Phrase Macros and XML IVRs
- Lua FreeSWITCH Scripting
- Dialplan in Deep
- Where are the previous Dialplan seasons?
- XML Dialplan Recap
- More about conditions
- Pitfalls to avoid
- Variables
- API commands from Dialplan
- FreeSWITCH string interpreter functions
- "Hotkeys", Listening, Barging
- Dialplan Cookbook
- Summary
- Dialplan, Directory, and ALL via XML_CURL and Scripts
- ESL - FreeSWITCH Controlled by Events
- HTTAPI - FreeSWITCH Asks Webserver Next Action
- Conferencing and WebRTC Video-Conferencing
- Handling NAT
- VoIP Security
- Troubleshooting, Asking for Help, and Reporting Bugs
ITSPs
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