5.2.2 SIP (Session Initiation Protocol) Extension
•
Before a SIP extension can be used with the PBX, the IP address of the mother board, password, and
extension number must be assigned on the SIP extension and on the PBX. Even if the IP terminal
registration mode has been set to full automatic mode or extension input mode, general SIP extensions
must be registered manually.
For details on how to register, refer to the Installation Manual.
•
When registering the SIP extension, the user ID must be the extension number of the SIP extension.
•
When a SIP extension uses the Call Hold feature, the target call is put on Consultation Hold.
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DSP Resource Usage
Making a call from a SIP extension requires a certain number of DSP resources, depending on the codec
used. If all DSP resources are in use, this operation cannot be performed. To ensure a minimum level of
performance, DSP resources can be reserved for VoIP communication. (→ 5.5.3 DSP Resource Usage)
However, DSP resources are not required for P2P calls. (→ 5.2.3 Peer-to-Peer (P2P) Connection)
Installation Manual References
4.4 Virtual Cards
PC Programming Manual References
9.17 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Card Property
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property
5.2.2.1 Simple Remote Connection
Description
Using the Built-in Media Relay Gateway
Using the Built-in Media Relay Gateway feature, KX-NT500 series IP-PTs, Panasonic IP-Softphones and
KX-NS0154 IP-CSs can be connected to a remote location without an additional device, such as an SBC
(Session Border Controller). These devices can be registered to the remote location after configuring the
Built-in Media Relay Gateway. For the KX-NSX Series and general SIP phones can also be installed to the
remote location easily.
Note
•
Peer-to-peer communication is not supported for the built-in media relay gateway.
Using an SBC (Session Border Controller)
KX-UT SIP phones and general SIP phones support simple remote connectivity when the KX-NSX Series is
networked with an SBC (session border controller). Simple remote connectivity means that even if the SIP
phone is located behind a NAT router, firewall, or both, specialised settings such as NAT traversal settings
do not need to be configured for each remote extension.
There are 2 scenarios for configuring and connecting a SIP phone:
1.
The SIP phone is connected and registered to the PBX on the PBX's local network. The necessary
settings are configured automatically by the PBX.
2.
The remote IP settings of the SIP phone are configured without first connecting the phone to the PBX.
Once programmed, the SIP phone is sent to the remote location, connected to the network and will
automatically connect to the PBX.
Conditions
•
Extensions that will be configured remotely use HTTPS for transferring the configuration file. However, a
maximum of 20 extensions per site can be connected in this way.
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The following types of settings must be configured on the PBX:
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Feature Manual