Setup Reference guide for Panasonic NS1000

ver2.1 “StarHub Pfingo” SIP Trunks

This document is a reference for configuring “StarHub Pfingo” SIP trunks onto Panasonic NS1000
and includes the settings required for Incoming Call DDI routing and Outgoing Call CLI presentation.

SIP trunk details such as account ID, authentication ID, password and DDI number will be provided
separately.

Attention:
This document was created based on the results of test environment, accounts.
Information of used in this document is for interoperability testing.
Information and Specifications in this document are subject to change without notice.
Also, we can not guarantee the SIP Trunking operations.
Please obtain the information from Service provider before setting of SIP trunk.
We are not responsible for any information in this document.

Version 1.0(PSY) 17.July, 2013

(1) Provisioning a SIP trunk: Page 2

(2) Incoming Call Routing: Page 6

(3) Outgoing Call CLI: Page 7

(4) How to use CLI presentation: Page 10

(1) Provisioning a SIP Trunk

SIP Trunk – Port Property

Important Note: Programming the details of the SIP trunk is done in this field.

Recommended setting:
- NAT - Keep Alive Packet Sending Ability change to “Enable”. (Default: Disable)

Move mouse over “V-SIPGW16” and click “Ous” --> click “OK” pop-up window.
After that, Move mouse over “V-SIPGW16” again and click “Port Property”.

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SIP Trunk – Port Property continued

[Main Tab]
1. Channel Attribute: Basic Channel
2. Provider Name: Enter the trunk a logical name
3. SIP Server Location – Name: sip.pfingo.com
4. SIP Server Location – IP Address: Not required
5. SIP Server port Number: Leave at 5060
6. SIP Service Domain: Not required
7. Subscriber Number: Not required

[Account Tab]
1. User name: Enter the SIP Account (User name) as supplied by Pfingo.
Please note that this is just the SIP Account (user name) and
DOES NOT include @sip.pfingo.com
2. Authentication ID: Enter the Authentication ID as supplied by Pfingo.
Please note that this is just the Authentication ID and
DOES NOT include @sip.pfingo.com
3. Authentication Password: Enter the Password as supplied by Pfingo.

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SIP Trunk – Port Property continued

[Register Tab]
1. Register Ability: Leave at Enable
2. Register Sending Interval: Leave at 3600
3. Un-Register Ability: Leave enabled
4. Registrar Server – Name: Not required
*If Register Server address is different from SIP proxy server, Enter the Register Server.
5. Registrar Server – IP Address: Not required
6. Registrar Server port number: Leave at 5060
7. Registrar Resending Interval: Leave at 300

Go Back to [Main] tab.

Drop down “Channel Attribute” and select “Additional channel for Ch1” for each further channel
that Pfingo have provided for this trunk.

Click “OK” to apply changes.

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SIP Trunk – Port Shelf property

Move mouse over “V-SIPGW16” again, and click “Shelf Property”.

NAT - Keep Alive Packet Sending Ability: Change to “Enable”

NAT - Keep Alive Packet Type: Confirm “Blank UDP”
NAT - Keep Alive Packet Sending Interval(s): Confirm “20”

Click “OK” to apply changes and click “Ins” (in Service).

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(2) Incoming Call Routing

Go to “CO & Incoming call” and select “3.DDI /DID Table”

1. DDI/DID Number:
2. DDI/DID Name:
3. DDI/DID Destination:
Enter the DDI number in the format 65+PSTN Number (as below)
Example: PSTN number=3000-9999
Enter: 6530009999
Determined by the installer (optional setting)
Determined by the installer (extension number, group etc)

All other settings can be left at default

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(3) Outgoing Call CLI

Important Note:
Specification of "From Header format" for SIP trunk of Starhub Pfingo. PBX's From header have to
set as following in order to notify the CLI to the Callee party. The use of other format in PBX can call,
but in this case will not notify the CLIP to callee.

From: "DDI Number" <sip: SIP Username (Authentication ID)@sip.pfingo.com>

For example, From: "6530009999" <sip:tr*test*sh_pfingo@sip.pfingo.com>

Go to 4.1.1 Extension Settings

Enter the “Extension Name” (this will be used in PBX local)
Select the “CO” to "CLIP on Extension/CO. and Click “OK” to apply changes.

Go to 10.1 CO Line Settings

Enter the “CO Name (CLI Number)” and Change the “Trunk Group Number (eg: 2)”

Click “OK” to apply changes.

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Outgoing Call CLI continued

Recommended settings
Go to “1.Slot - Virtual”

Move mouse over “V-SIPGW16” and click “Ous” --> click “OK” pop-up window.
After that, Move mouse over “V-SIPGW16” again and click “Shelf Property”.
SIP Called Party Number Check Ability: Select the value to "Disable (Low->High)"

Click “OK” to apply changes and click “Ins” (in Service).

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Go to Extension Settings
Configure the COS Settings (Use the COS, If necessary)

9
(4) How to use CLI Presentation.

[Method-1]

Use the "Trunk Line Access" way. (Example of use)

Trunk Group number 01: SIPGW Port-1
Trunk Group number 02: SIPGW Port-2

8 + 01 + XXXXXXXX (Test number: eg 8-01-3100-2222 from Extension 201)

--> Callee Party will be presented eg 6530009999 as CLI.

8 + 02 + XXXXXXXX (Test number: eg 8-02-3100-2222 from Extension 202)

--> Callee Party will be presented eg 6530001111 as CLI.

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[Method-2]

Use the “COS of External Call Block”. (Example of use)

Extension201 use the COS-1, and Extension 202 use the CoS-2.
Calls from Extension 201
0 + XXXXXXXX (TRG is used 01, eg test number: 0-3100-2222)
--> Callee Party will be presented eg 6530009999 as CLI.

Calls from Extension 202

0 + XXXXXXXX (TRG is used 02, eg test number: 0-3100-2222)
--> Callee Party will be presented eg 6530001111 as CLI.

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