Click on the New Sip Trunk to go to the Edit Sip Trunk page.
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- NAME: a meaningful name for this trunk
- HOST: IP address/hostname of the SIP server provided by ITSP
- PORT: this is 5060 by RFC
- TRANSPORT: UDP
- ENCRYPTION: disabled (unchecked)
We do also suggest the following values to be set:
- AUDIO TONES: ON EARLY MEDIA works fine with the Cisco Unity Call Manager. Avaya instead can deal with both ON EARLY MEDIA and ON ANSWERED CALLUnified 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.
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