Table 50: Create New Sip Trunk - Grandstream Networks UCM6102 User Manual

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Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.
Click on
to configure detailed parameters for the VoIP trunk.
Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk.
Click on
to start LDAP Sync.
Click on
to delete the VoIP trunk.
For VoIP trunk example, please refer to the document in the following link:
http://www.grandstream.com/products/ucm_series/ucm65xx/documents/how_to_interconnect_multiple_uc
m6510_using_peer_trunk.pdf
The VoIP trunk options are listed in the table below.
Type
Provider Name
Host Name
Keep Original CID
Keep Trunk CID
NAT
Disable This Trunk
TEL URI
Firmware Version 1.0.9.26

Table 50: Create New SIP Trunk

Select the VoIP trunk type.
Peer SIP Trunk
Register SIP Trunk
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider's server of the
trunk.
Keep the CID from the inbound call when dialing out. This setting will
override "Keep Trunk CID" option. Please make sure that the peer PBX at
the other side supports to match user entry using "username" field from
authentication line.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
Turn on this setting when the PBX is using public IP and communicating
with devices behind NAT. If there is one-way audio issue, usually it is
related to NAT configuration or SIP/RTP port support on the firewall.
If checked, the trunk will be disabled.
Note:
If a current SIP trunk is disabled, UCM will send UNREGISTER message
(REGISTER message with expires=0) to the SIP provider.
If the trunk has an assigned PSTN telephone number, this field should be
UCM6100 Series IP PBX User Manual
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