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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. 

    NoteNo 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
    • both ON EARLY MEDIA
    and 
    •  and ON ANSWERED CALL.
    • 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
    • DISALLOW: the values of this fields highly depends from what the other end is capable of. Our suggestion is to leave this field empty

    • 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.

    2.1.1 Certificate Management 

    sdgsdg

    ...