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Click on the New Sip Trunk to go to the Edit Sip Trunk page.

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

In 

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AnchorNameedit sip trunk form
 you can read an example configuration for an unencrypted SIP trunk. Mandatory fields for the unencrypted configuration are:

  • NAME:
  • a
  •  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

Optional fields are:

  • VIRTUAL PHONE NUMBER: associated virtual number
  • NAT: enable/disable the Network Address Translation configurationthis is 5060 by RFC
  • TRANSPORT: UDP
  • ENCRYPTION: disabled (unchecked)

We do also suggest the following values to be set:

  • AUDIO TONESON 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.

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