Sip Trunk For Home Full Example Reference#

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Sip Trunk For Home Full Example Reference

A working installation of Asterisk, preferably with one or more telephones configured and working, that can dial the Asterisk server, and the other phones connected to it successfully. Full example reference configuration filesBefore we start there are a couple of things that we need: Have a look here for some alternatives.When you have bought a suitable SIP trunk, and have gotten your account information from the provider, we can continue, and set it up. Choose another one instead. Everyone wants to sell a service, nothing wrong with that, but watch out for VOIP-providers that explicitly filter connections from users own Asterisk-servers. Some companies don’t want people to run their own PBXs and create their own services, for free, and with the freedom that comes with using Asterisk, they want you to pay for their services, like voicemail, “answering machines”, and such.

I will continue where the previous article left off, and use the configuration files that was created there, and add a SIP trunk to this setup, step by step.Edit sip.conf in your favourite editor and add the following example configuration: Register and get calls from Foo Provider, to our number 1-55Register => prefer to have the above sections at the top in my sip.conf, but that is up to you. For a basic configuration only two files needs to be edited, sip.conf and extensions.conf. A SIP trunk, with a provider of your choice.Setting up a SIP trunk is not harder than adding a SIP telephone.

Context=incomingNotice that we send all incoming calls to a specific, and named part of our dialplan. We have to register to be able to have calls to our telephone number be forwarded to us. Some notes about the above configuration: register => register directive registers our Asterisk with the trunk-providers SIP-server, with the username ( 15554551337 in our example case) and the password ( password123), that we have specified.

Dial your Asterisk server from your mobile phone, and hopefully your first SIP telephone will ring. To reload the SIP configuration and the dialplan, connect to the running Asterisk’s command-line: asterisk -vcrVerify that your Asterisk server registers with your provider correctly: sip show registryIf necessary, troubleshoot the registration, use the following Asterisk CLI commands: sip set debug onNow at last, test the configuration. You also exchange “fooprovider” with the name of your real provider that you configured in sip.conf.For our configuration to take effect we either have to reload it from Asterisk’s command-line interface, or restart Asterisk. “60” is the number of seconds to let it ring, until we give up and let Asterisk play congestion tones to us, increase the time value if you think it is too short. Edit extensions.conf, and add: Exten => s,1,Log(NOTICE, Incoming call from $” uses the entered number, after the first digit, that is the meaning of “:1”.

Replace the IP address and mask below with the actual IP address and mask of the computer running the softphone, or the address of the hardware phone, either a host address and full mask, or a network address and correct mask, registering will be allowed from that host/network. Register and get calls from Foo Provider, to our number 1-55Register => - Otherwise we would define the IP address or FQDN of the phone on the following line.