In this lesson we will look at SIP providers. Using a SIP provider, the Asterisk server can connect to a company that can provide a phone number and handle outside phone calls.
There are many companies that server as SIP providers. Using the SIP protocol the Asterisk server can register to the provider, similar to adding a SIP peer in the previous lessons, and a dialplan for that provider can be created. SIP uses the IP protocol and can be used to connect to servers over the Internet.
If you type in “Asterisk SIP providers” in a search engine you can find many different companies. This lesson does not give any recommendation on SIP providers, as each one of the companies provides various advantages. Because students of this course are coming from all over the world, a website that is a good resource to get started reviewing providers by country is https://www.voip-info.org/wiki/view/SIP/IAX+Services+for+Asterisk (Links to an external site.)
Some things to consider are call quality, codecs, and pricing. Call quality will depend on your Internet connection. Troubleshooting call quality can be difficult. Some providers can give you an Internet connection and a separate connection for voice, which is often more expensive. Make sure your provider has the same codecs you are using on your Asterisk.
One last thing to point out is the difference between a SIP peer and trunk. If the provider offers a SIP trunk, that connection can carry multiple phone numbers, known as DIDs (Direct Inward Dialing). A SIP peer will carry just has a single DID number. A SIP peer will be used in this section.
The SIP provider QuestBlue.com (Links to an external site.) will be used in this section to demonstrate how to register the Asterisk server and configure incoming/outgoing calls. There is no affiliate connection with QuestBlue or recommendations being offered. They happen to be a highly rated provider with international connections.