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UCM6200 Series User Manual
Version 1.0.19.27
VOIP TRUNKS
VoIP Trunk Configuration
VoIP trunks can be configured in UCM6200 under Web GUIExtension/TrunkVoIP Trunks. Once created,
the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the
trunk.
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/sites/default/files/Resources/ucm6xxx_sip_trunk_guide.pdf
The VoIP trunk options are listed in the table below.
Table 51: Create New SIP Trunk
Type
Select the VoIP trunk type.
Peer SIP Trunk
Register SIP Trunk
Provider Name
Configure a unique label (up to 64 character) to identify this trunk when listed in
outbound rules, inbound rules etc.
Host Name
Configure the IP address or URL for the VoIP providers server of the trunk.
Keep Original CID
Keep the CID from the inbound call when dialing out. This setting will override
“Keep Trunk CIDoption. Please make sure that the peer PBX at the other side
supports to match user entry using “usernamefield from authentication line.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when the
extension has CID configured. The default setting is “No”.
NAT
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.