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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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 44xxxxxxxxxx@spitfiretsp.net. 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:44xxxxxxxxxx@spitfiretsp.net

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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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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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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.