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Configuring Panasonic NCP 4.1001 for Spitfire SIP Trunks (with
no individual CLI)
This document is a guideline for configuring Spitfire SIP trunks
onto Panasonic NCP 4.1001 and includes the settings required for
Inbound DDI routing and Outbound CLI presentation. The settings
contained within have been tested and are known to work at the time
of testing. SIP trunk details such as account number and password
will be provided separately.
Provisioning a SIP trunk: Page 2 Inbound Routing: Page 7
Outbound Routing - Reference only: Page 8 Outgoing CLI: Page 9
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2
Provisioning a SIP Trunk Add SIP Trunk – Port Property Important
Note: Programming the details of the SIP trunk is done in this
field. In this example the system has been programmed to use the
Username (Account tab) and SIP Service Domain (Main tab) to
populate the From URI for authentication (this is required for
registration & to allow outbound calls to be authorized by the
SIP trunk server) Alternatively you can enter these details in the
From Header, SIP-URI (Calling Party tab) in the format
[email protected]. Please be aware that entering details
in the From Header, SIP-URI will over-ride the Username and SIP
Service Domain. Right click on “V-SIPGW16” and click on “Port
Property”.
mailto:[email protected]
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Add SIP Trunk – Port Property continued Main Tab
1. Channel Attribute: Basic Channel 2. Provider Name: Give the
trunk a logical name 3. SIP Server Location – Name:
mproxy3.spitfiretsp.net – this will be used if DNS is
available 4. SIP Server Location – IP Address: 83.218.143.16 –
not required if using DNS 5. SIP Server port Number: Leave at 5060
6. SIP Service Domain: spitfiretsp.net 7. Subscriber Number: Not
required
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4
Add SIP Trunk – Port Property continued Account Tab
1. User name: Enter the User name as supplied by Spitfire.
Please note that this is just the number and DOES NOT include
@spitfiretsp.net
2. Authentication ID: Enter the User name as supplied by
Spitfire. Please note that this is just the number and DOES NOT
include @spitfiretsp.net
3. Authentication Password: Enter the password as supplied by
Spitfire
mailto:@spitfiretsp.netmailto:@spitfiretsp.net
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Add SIP Trunk – Port Property continued Register Tab
1. Register Ability: Leave at Yes 2. Register Interval: Leave at
3600 (valid range is 300 to 3600) 3. Un-Register Ability: Leave
enabled 4. Registrar Server – Name: mproxy3.spitfiretsp.net – this
will be used if DNS is available 5. Registrar Server – IP Address:
83.218.143.16 – not required if using DNS 6. Registrar Server port
number: Leave at 5060
All other tabs may be left at default
- Option - Called Party - Calling Party - Voice/Fax - RTP/RTCP -
T.38 - T.38 Option - DSP - Supplementary
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6
Add SIP Trunk – Port Property continued Go back to “Main” tab
Drop down “Channel Attribute” and select “Additional channel for
Ch1” for each further channel that Spitfire have provided for this
trunk. Click “OK” or “Apply” to apply changes and put the card In
Service.
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Inbound Routing Go to “CO & Incoming call” and select “3.DDI
/DID Table”
1. DDI/DID Number: Enter the DDI number in the format
44xxxxxxxxxx (as below) 2. DDI/DID Name: Determined by the
installer (optional setting) 3. DDI/DID Destination: Determined by
the installer (extension number, group etc)
All other settings can be left at default
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Outbound Routing with ARS – Reference only There are many
different ways to program Outbound Routing and many companies have
their own method of implementing it. Even our own ARS programming
requirements change from location to location and site to site. For
this reason I have not included any ARS programming in this guide
One recommendation however, is that you reduce the Extension
Inter-digit timer. This is the not only the delay allowed between
dialing digits but is also how long a dialed call will wait if it
does not find an exact number length match in the ARS table. If you
do not use any ARS digit length matching then every call will wait
for this timer to expire after the last digit has been dialed
before sending the call to line. This may become a little
frustrating for the End User so it may be worthwhile to lower this
timer. Go to “2.System, 3.Timers & Counters”
Outgoing CLI By default the SIP trunk is provisioned to present
the main number only on outgoing calls. If you would like the
ability to present individual outgoing CLI please contact the SIP
co-ordination manager on 020-7501-3016.