How to Create a SIP Trunk in AsteriskNOW

Written by alexander gokhfeld
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How to Create a SIP Trunk in AsteriskNOW
Computer-based phone calls use SIP protocol. (Dynamic Graphics/Dynamic Graphics Group/Getty Images)

Session Initiation Protocol (SIP) trunk services are available from the Voice-over-IP (VoIP) providers. They allow you to make VoIP phone calls from tyour computer without installing any additional hardware devices. The AsteriskNOW software uses the SIP trunks to connect to other telephone services including tpublic switched telephone networks and VoIP services. You need the AsteriskNOW configuration with the trunking settings of the VoIP provider to use SIP trunk in the AsteriskNOW software.

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  1. 1

    Connect to the FreePBX by going to http:// [IP address of your SIP provider] /admin/.

  2. 2

    Click the "OK" button in the login window to accept the user name "freebox" and the password "fpbx."

  3. 3

    Select the "Add SIP Trunk" from the "Trunks" tab.

  4. 4

    Enter your SIP telephone number in the "Outbound Caller ID" box.

  5. 5

    Select the "Never Override CallerID" check box.

  6. 6

    Enter the name of your SIP provider in the "Trunk Name" box of the "Outgoing Settings" section.

  7. 7

    Enter the following settings for your SIP Trunk account in the "PEER" box :

    username = "Your SIP Trunk Login ID"

    type = peer

    secret = "Your SIP Trunk Authorization Password"

    insecure = very

    host = eps1."SIP provider name".net

    dtmfmode = rfc2833

    allow = ulaw

  8. 8

    Remove the default settings in the "User Details" box of the "Incoming Settings" section.

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