Click on the New Sip Trunk to go to the Edit Sip Trunk page.
Subtitle | ||||||
---|---|---|---|---|---|---|
| ||||||
In
Xref | ||
---|---|---|
|
- 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 TONES: ON EARLY MEDIA works fine with the Cisco Unity Call Manager. Avaya instead can deal with both ON 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
- 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.
Navbar |
---|