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Subtitle
Prefixfigure
SubtitleText"Edit Sip Trunk" form
AnchorNameedit sip trunk form

In 

Xref
AnchorNameedit sip trunk form
 you can see an example configuration for creating a SECURE Inbound SIP Trunk. The mandatory values are:

  • NAME: a meaningful name for this trunk
  • HOST: IP address/hostname of the SIP server provided by ITSP
  • PORT: this is 5061 by RFC
  • TRANSPORT: TLS
  • ENCRYPTION: check it enabled
  • DIRECTMEDIA: check it enabled
  • SENDRPID: check it enabled

The default values for Port, Allow and Disallow are usually work well, however feel free to change them to suite your needs better.

Optional fields are:

  • CONCURRENT CALLS: you can leave the default, which is often a practice value for the day-by-day service and change it later if you notice call drops on this Trunk.

  • NAT: check this box only if you are not using directly the public IP address on the trunk interface. 

    Note

    No interface binding for the trunk! The trunk would use the default gateway you configured in the PSAM 2.1 Network Configuration

    • We do also suggest the following values to be set:

    • AUDIO TONES: ON EARLY MEDIA
    works
    •  works fine with the Cisco
    Unity Call Manager. Avaya instead can deal with both N EARLY MEDIA and ON ANSWERED CALLDISALLOW: the values of this fields highly depends from what the other end is capable of. Our suggestion is to leave this field empty
    • Unified Communications Manager.
    • DTMFMODE: choose your values considering the PBX on the other end of the Trunk. Usually we suggest to choose the value RFC2833
  • ALLOW: the values of this fields highly depends from what the other end is capable of. Our suggestion is to leave the default: g729,amr,gsm,ulaw,alaw
    • DIRECTMEDIA: enabled (checked)
    • SENDRPID: enabled (checked)

    Other fields in the form depend by your network topology and by the features on the other end PBX.

    When you are done with your changes, commit them by clicking on the Update icon.

    4.1.1 Certificate Management 

    In order to validate a TLS peer for establishing an Encrypted SIP Trunk, you generally have to import the other party CA Root. This is important because the peer TLS certificate could not match PrivateServer actual Certificate Chain and thus the validation would fail.

    Info

    PrivateServer comes with a bundle of the most known CA Roots certificates ready to be used. So this section is useful for minor certificate authorities and/or for self signed certificates.

    Please read 2.3.3 Add certification authority to understand how to import a new CA Root certificate.

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