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• 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
• this is 5061 by RFC
• TRANSPORT: pro
• 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.

Mandatory Optional 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 offerCONCURRENT 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 if you are not using directly the public IP address.
• AUDIO TONES: ON EARLY MEDIA works fine with the Cisco Unity Call Manager. Avaya instead can deal with both N EARLY MEDIA 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.