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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 fields values are:

  • NAME: a meaningful name for this trunk
  • HOST: IP address/hostname of the SIP server provided by ITSP
  • PORT: registration port of the service
  • CONCURRENT CALLS: maximum number of calls on a single trunk
  • DIRECTMEDIA: experimental sound management in P2P mode without routing the audio stream through the server
  • SENDRPID: the Remote Party ID, used to interconnect with VoIP Provider for the management of privacy
  • DTMFMODE: tone signaling technology used for this SIP interconnection
  • ALLOW: voice codecs used in this trunk
  • DISALLOW: voice codecs denied in this trunk

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

Mandatory fields are:

  • NAT: enable/disable the Network Address Translation configuration
    • AUDIO TONES: this is used for Audio Messaging. The possible values are:
  • NO AUDIO TONES: no audio messages would be played on this trunk
  • ON EARLY MEDIA: the audio messages would be played using SIP early offer
    • ON ANSWERED CALLS: the audio messages would be played using SIP delayed offer
    • this is 5061 by RFC
    • TRANSPORT: TLS
    • ENCRYPTION: check it enabled

    We do also suggest the following values to be set:

    • AUDIO TONES: ON EARLY MEDIA works fine with the Cisco 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
    • 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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