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SIP TRUNKING PANASONIC KX-NCP/KX-TDE: ADMINSTRATOR GUIDE 1.1 Author: BTW Product Line Date: 18/05/2011

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Page 1: SIP TRUNKING - Rainbow Communications Your · PDF fileSIP TRUNKING -NCP/KX TDE: ... They use the Session Initiation Protocol (SIP) ... SIP Trunk Users and SIP Trunk Mobility Users

SIP TRUNKING PANASONIC KX-NCP/KX-TDE: ADMINSTRATOR GUIDE 1.1

Author: BTW Product Line

Date: 18/05/2011

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Confidentiality

All information in this document is provided in confidence as per your terms and conditions

with BT and shall not be published or disclosed wholly or in part to any other party without BT’s prior permission in writing and shall be held in safe custody. These obligations shall not apply to information which is published or becomes known legitimately from some source other than BT.

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Document Revision History

Release Version Reason for Change Date Author

1 1.0 Document Created 25/02/2011 BTW Product Line

1 1.1 Document Updated

Updated incorrect instances of uk.ic.sipconnect.IPVS.co.uk to uk.ic.sipconnect.hipcom.co.uk

18/05/2011 BTW Product Line

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CONTENTS

DOCUMENT REVISION HISTORY ................................................................................................ 3

1 INTRODUCTION ................................................................................................................ 6

2 OVERVIEW ....................................................................................................................... 7

3 DEVICE CAPABILITIES AND KNOWN INTEROPERABILITY ISSUES ........................................... 8

4 SOLUTION CONFIGURATION .............................................................................................. 9

5 CONNECTING TO THE PANASONIC IP-PBX ......................................................................... 10

6 PROVISIONING ............................................................................................................... 11

7 PANASONIC IP-PBX CONFIGURATION ............................................................................... 12

7.1 SIP Settings ............................................................................................................................................. 13 Shelf Property.......................................................................................................................................................... 13 Card Property .......................................................................................................................................................... 14 Port Property ........................................................................................................................................................... 14

7.2 Trunk Group Settings ............................................................................................................................... 23

7.3 Dialing Plan ............................................................................................................................................. 24

7.4 Extension Settings ................................................................................................................................... 25

7.5 CO Line Settings ....................................................................................................................................... 26

7.6 DIL Table & Port Settings .......................................................................................................................... 27

7.7 DDI / DID Table ........................................................................................................................................ 28

7.8 COS Settings ............................................................................................................................................ 29

7.9 System Options ........................................................................................................................................ 30

8 TROUBLESHOOTING AND USEFUL INFORMATION ............................................................. 33

8.1 Checking SIP Messaging To/From the PBX .................................................................................................. 33

8.2 Device Fails to Register ............................................................................................................................ 34

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8.3 Outbound Calls Fail .................................................................................................................................. 35

8.4 Inbound Calls Fail ..................................................................................................................................... 36

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1 INTRODUCTION

This document describes the configuration procedures required for the Panasonic KX-NCP and KX-

TDE IP-PBX ranges to make full use of the capabilities of IPVS SIP Trunking. These ranges of Panasonic IP-PBX’s are some of many access products that interoperate with IPVS. They use the Session Initiation Protocol (SIP) to communicate with IPVS for call control.

This guide describes the specific configuration items that are important for use with IPVS. It does not describe the purpose and use of all configuration items on the KX-NCP and KX-TDE ranges. For those details, see the relevant Panasonic documentation.

Interoperability testing validates that the device interfaces properly with IPVS via the SIP interface.

Qualitative aspects of the device or device capabilities not affecting the SIP interface such as display

features, functionality and performance are not covered by interoperability testing. Requests for

information and/or issues regarding these aspects should be directed to manufacturer.

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2 OVERVIEW Please note that the following prerequisites must be satisfied for successful deployment: Any SIP call must present a number with a current registration on the IPVS platform in one or more of

the following headers of the SIP message transmitted by the IP-PBX (listed in order of significance): 1. Diversion 2. P-Asserted-Identity 3. Remote-Party-Id

4. From If you see a 604 SIP message back from IPVS then the device is not presenting the correct CLI to the IPVS platform or the account is not present on the IPVS platform.

SIP Trunking DDI Users must have an active Public Number (DDI) on the IPVS Platform. Legacy PBX users/extensions (who do not have a SIP Trunking User account provisioned) must present the SIP Trunk Group number or ‘Bearer Number’ as their outbound CLI for calls to be able to traverse the IPVS platform.

NOTE:

Please see our Authorised Equipment List to check the current available recommended software/firmware version. It is strongly recommended to

use the current supported software/firmware. The Supported Device List is available from the Support Centre

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3 DEVICE CAPABILITIES AND KNOWN INTEROPERABILITY ISSUES

Comments

Device Type IP-PBX

Verified Revisions KX-NCP500/1000 – Fw. 4.1

KX-TDE100/200/600 – Fw. 4.1

IPVS R14SP9

Route Using TO Header Yes

SIP Connect Yes

Registration Capable Yes

SIP Proxy/Registrar uk.ic.sipconnect.hipcom.co.uk

Outbound Proxy uk.ic.sipconnect.hipcom.co.uk

Codecs G711u/G711a/G729a

RFC 2833 DTMF Yes

T.38 Fax Yes Not tested

Call Control

- Hold - Call Waiting

- Blind Transfer

- Consultive Transfer - Three Way Call

Yes

Yes

Yes Yes

Yes

Call Services

- PBX Redirect - IPVS CLI Blocking

- IPVS Call Pickup

Partial

Yes

Yes

Redirect by PBX to IPVS Voicemail

is not supported

IPVS Enhanced Features

IPVS Voicemail / Local Voicemail Both

Voicemail MWI No Use IPVS Voicemail to Email or Email Notification (Call Notify)

SIP Trunk Mobility User Features

- Call Forward Always

- Call Forward Busy - Call Forward No Ans

- Call Forward Unreach

- Call Notify - Intercept User

- SIM Ring

- Remote Office

Yes Yes

Yes

Yes Yes

Yes

Yes Yes

Set via Customisable Feature

Access Codes, Business Portal or

Toolbar

Toolbar

- Click-to-Dial

- Call Control - Services

Yes

Yes Yes

Yes

Hold on PBX handset and Toolbar must be used together

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4 SOLUTION CONFIGURATION

The following diagram shows an example of a typical deployment configuration with the Panasonic KX-NCP or KX-TDE. Typically, the IP-PBX is placed at the customer premises with either Analogue to IP handset connected/registered to it.

The IP-PBX registers the main line (or Trunk Group number) with the Trunk Group configured on IPVS. This enables all PBX users to be implicitly registered with IPVS via the trunk group number registration.

IP Network

Panasonic

KX-NCP or

KX-TDE

Trunk Group Number

[email protected]

Trunk Group Profile Data:

Device Type: Panasonic KX-NCP or Panasonic KX-TDE

Trunk Group Number Profile Data:

Line/port: [email protected]

Authentication Username: 441234567890

Authentication Password: password

DDI: 01234 567890

PBX User 1:

User Type: Business Trunking User

Feature Package: SIP Trunk User / SIP Mobility User etc

Add-ons: Toolbar, Voicemail etc

Device Type: Trunk Group

Line/port: [email protected]

DDI: 01234 567891

PBX User 2:

User Type: Business Trunking User

Feature Package: SIP Trunk User / SIP Mobility User etc

Add-ons: Toolbar, Voicemail etc

Device Type: Trunk Group

Line/port: [email protected]

DDI: 01234 567892

SIP Registra/Proxy Data:

FQDN: uk.ic.sipconnect.hipcom.co.uk

Port: 5060

Outbound Proxy Data:

FQDN: uk.ic.sipconnect.hipcom.co.uk

Port: 5060

Router / IP

access

PBX User 1

01234 567891

PBX User 2

01234 567892

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5 CONNECTING TO THE PANASONIC IP-PBX

To connect to the Panasonic IP-PBX install the manufacturer recommended PBX Unified Maintenance Console, for details please follow the associated documentation.

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6 PROVISIONING

Before reading and using this Admin Guide you must read the SIP Trunking Provisioning Administrator Guide that can be found on the Support Centre: Support Centre >> Downloads >> SIP Trunking >> SIP Trunking Provisioning Admin Guide

NOTE:

The SIP Trunking Provisioning Admin guide details how to provision SIP Trunk Groups, SIP Trunk Users and SIP Trunk Mobility Users. This guide

assumes that you have read this guide and the required provisioning has been completed.

When provisioning please select the appropriate Shared Device Type:

Panasonic KX-NCP or Panasonic KX-TDE

(Dependant on your IP-PBX type)

Note: If you do not have this device type available in your service offering please contact your Account Manager who will be happy to arrange its addition for you.

We would also recommend reading the SIP Trunking Overview Guide that can be found on our Support Centre for background on the messaging that a SIP Trunking device should present to the IPVS platform to be able to successfully make calls.

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7 PANASONIC IP-PBX CONFIGURATION

The following steps show how to program a Panasonic KX-NCP to KX-TDE to interconnect with IPVS using SIP Trunking. The steps in this section assume that you are connected to the relevant PBX with Installer privileges

using Panasonic’s PBX Unified Maintenance Console. They also assume that you have prior knowledge of the Maintenance Console and the setup of a Panasonic PBX using it. For more details on the PBX Unified Maintenance Console and configuration of a Panasonic PBX please consult the relevant Panasonic documentation.

NOTE:

Before installing and configuring the Panasonic IP-PBX, ensure that you read the Panasonic Safety and Compliance Information and any other related Panasonic

documentation. The configuration steps discussed in this section have been taken from an example configuration of a Panasonic KX-NCP500. The steps are applicable to all KX-NCP and KX-TDE PBX’s discussed in the Device Capabilities and Known Interoperability Issues

section of this document.

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7.1 SIP Settings

Configuration > Slot > IPCMPR Virtual Slot > V-SIPGWXX This section assumes that you have installed and enabled the virtual SIPGW card that is installed in your PBX.

Before carrying out any of the following configuration steps it is recommended that you take the V-SIPGW card out of service. Once all steps have been carried out the card should be brought back in to service.

Shelf Property

Select the Shelf Property link for the V-SIPGW card. Main Tab

Under the Main tab the following important values should be set: SIP Client Port Number 5060

NAT Traversal Off/Fixed IP Addr. (Dependent on network setup), STUN should not

be enabled.

NAT – Voice (RTP) UDP Port No. 16384

STUN Ability Disable

SIP Called Party Number Check Ability Disable(Low->High)

Symmetric Response Routing Ability Enable

100rel Ability Enable(Passive)

Ringback Tone to Outside Caller Disable

The below image shows the example settings for our solution.

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Timer Tab

The default values under this tab do not need to be changed.

Click OK once you have finished entering the necessary settings.

Card Property

Select the Card Property link for the V-SIPGW card. The values configurable under this section are mainly dependent on the setup of the LAN that the PBX will be a part of.

We recommend that under these settings DNS SRV Record Resolve Ability is disabled as this is not currently supported by the IPVS platform. Click OK once you have finished entering the necessary settings.

Port Property

Select the Port Property link for the V-SIPGW card.

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In our example the card available is a V-SIPGW16 meaning it has 16 configurable channels each able to support a single SIP call. When configuring channels for connections to IPVS it is only necessary to configure the first channel and any further required channels should be selected as additional channels to this first channel. The steps for configuring this are discussed in this section.

Main Tab

Under the Main tab the following important values should be set for your desired primary channel: Channel Attribute Basic Channel

Provider Name We recommend setting this as a meaningful value e.g. IPVS

SIP Server Location - Name uk.ic.sipconnect.hipcom.co.uk

SIP Server Location – IP Address This is only required if the PBX cannot access a DNS server, do not enter unless necessary; 85.119.63.4

SIP Server Port Number 5060

SIP Service Domain uk.ic.sipconnect.hipcom.co.uk

Subscriber Number Trunk Group DDI as provisioned in the Business Portal e.g. 441234567890

Part of the configuration for our example setup is shown below.

Account Tab

Under the Account tab the following important values should be set for your desired primary channel: Username Trunk Group DDI as provisioned in the Business Portal e.g.

441234567890

Authentication ID Trunk Group Username as provisioned in the Business Portal

Authentication Password Trunk Group Password as provisioned in the Business Portal

The configuration for our example setup is shown below.

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Register Tab

Under the Register tab the following important values should be set for your desired primary channel: Register Ability Enable

Register Sending Interval (s) 3600

Un-Register Ability when port INS Enable

Registrar Server - Name uk.ic.sipconnect.hipcom.co.uk

Registrar Server – IP Address This is only required if the PBX cannot access a DNS server, do not

enter unless necessary; 85.119.63.4

Registrar Server Port Number 5060

The configuration for our example setup is shown below.

NAT Tab

No values should be entered under this tab as STUN is not used. Option Tab

Under the Option tab the following important values should be set for your desired primary channel:

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Session Timer Ability Enable(Active)

Session Expire Timer (s) 900

Refresh Method re-INVITE

Proxy-Require Option Leave Blank

The configuration for our example setup is shown below.

Calling Party Tab

Under the Calling Party tab the following important values should be set for your desired primary channel: Header Type P-Preferred-Identity Header

From Header – User Part PBX-CLIP – Setting this will allow you to present either the Trunk Group or Trunk User DDI for outbound calls on a per user basis.

From Header – SIP-URI Leave Blank

P-Preferred-Identity Header – User

Part

This should match what you have entered in the From Header –

User Part field.

P-Preferred-Identity Header – SIP-

URI

Leave Blank

Number Format National

Remove Digit 0

Additional Dial Leave Blank

Anonymous format in “From” header Display name and SIP-URI

The configuration for our example setup is shown below.

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Called Party Tab

Under the Called Party tab the following important values should be set for your desired primary channel: Number Format National

Type To header

The configuration for our example setup is shown below.

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Voice/FAX Tab

Under the Voice/FAX tab the following important values should be set for your desired primary channel, the chosen values should then be replicated across all channels that you will use: IP Codec Priority The 3 priority boxes allow you to specify the preferred codec order

for the PBX, in general we recommend;

1st

– G.711A 2

nd – G711Mu

3rd

– G.729A

If you would like to implement a G.729A only solution due to

bandwidth constrains then select G.729A as the first priority and

None for the remaining 2 options.

Packet Sampling Times (s) These values should be set as follows for each codec;

G.711A – 20ms G.711Mu – 20ms

G.729A – 30ms

Voice Activity Detection for G.711 Disable

Reserved Disable

Inform Annex B Status (G.729A) Enable

Fax Sending Method G.711 Inband or T.38 depending on your preference

Maximum Bit Rate Dependant on your preference

FAX Detection Ability Enable if you wish to support Fax

DTMF Outband (RFC2833)

Payload Type 101

Reserved 100

The configuration for our example setup is shown below.

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RTP/RTCP Tab

Under the RTP/RTCP tab the following important values should be set for your desired primary channel, the chosen values should then be replicated across all channels that you will use: RTP QoS Ability DSCP – This is our recommended methodology, although this DSCP

flag must be maintained through the access network to be useful

RTP QoS-DSCP 46 (EF)

RTCP Packet Sending Ability Enable

RTCP Packet Interval 5s

The configuration for our example setup is shown below.

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T.38 Tab

If you wish to fax through the PBX using T.38 then we would advise leaving the values in this tab as the defaults. T.38 Option Tab

If you wish to fax through the PBX using T.38 then we would setting the QoS method to DSCP and

setting the QoS-DSCP value to 46 (EF) as for the RTP setting. DSP Tab

We would advise leaving these values as the defaults unless advised otherwise.

Supplementary Service Tab

Under the Supplementary Service tab we recommend setting the following values for all used channels: CLIR Yes

CNIP (Send) Yes

CNIP (Receive) No

The configuration for our example setup is shown below.

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Assigning Additional Channels

Once you have entered all of the settings in the tabs as discussed it is necessary to go back to the Main tab and associate as many channels as are required to the primary channel that has been configured.

In the Channel Attribute drop down box select the Additional channel for option that references the channel you have configured. An example for out configuration where 4 channels are used is shown below.

Once this has been complete press OK to confirm the settings and apply them. We also recommend that after each section you back up the settings to the SD card using the relevant option under the tools menu of the Maintenance Console.

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7.2 Trunk Group Settings

Group > Trunk Group > TRG Settings Main Tab

Under the Main tab we recommend that the values are set as follows. In our example we only have one Trunk Group setup so we use Trunk Group 1.

Group Name We recommend setting this as a meaningful value e.g. IPVS

COS We recommend that the same COS group is used for all devices on

the PBX

CO-CO Duration Time (*60s) This controls the maximum duration of a trunk to trunk call so we

recommend setting it to none

Extension-CO Duration Time (*60s) This controls the maximum duration of an extension to trunk call so

we recommend setting this to none

Dialling Plan Table This should match the dial plan that you configure

The configuration for our example setup is shown below.

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7.3 Dialing Plan

Group > Trunk Group > Dialling Plan The dial plan that you enter is relative to your local area and the calls that you want to allow you r users to make. We recommend that you consider the following dialling options:

Star (*) Feature Access Codes Hash (#) Feature Access Codes International Calls 6 Digit Local Calls / 7 Digit Local Calls

118 Enquiries Calls 999 Emergency Calls 112 Emergency Calls 123 Emergency Calls

The below image taken from our solution shows some example rules from our proposed solution, it does not cover all dialling options.

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7.4 Extension Settings

Extension > Extension Type > Extension Settings The settings that you enter here are dependent on your setup and the phone types that you are using as well as the user’s preferences.

The important value within the settings is the ISDN CLIP – CLIP DDI (the value that the phone presents as the From number to the IPVS platform). This should be in the format of the DDI set in the Business Portal for the Trunk Group/User i.e. 441234567890.

If you just want to present the Trunk Group DDI for all users then enter this DDI in the CLIP DDI field for each user otherwise you can enter the Trunk User DDI for each user or a mixture of the two. The example image below is for Wired Extensions and each user is presenting their own DDI.

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7.5 CO Line Settings

CO & Incoming Call > CO Line Settings The CO Line Settings page allows you to assign the different virtual SIP channels/ports on the PBX to a desired Trunk Group and label them.

CO Name We recommend setting this as a meaningful value e.g. IPVS Channel

1

Trunk Group Number This should match the Trunk Group you setup previously.

The configuration for our example setup is shown below.

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7.6 DIL Table & Port Settings

CO & Incoming Call > DIL Table & Port Settings We recommend that you used the DDI/DID Distribution Method for inbound calls, to make sure that this is the case set all used channels to this as shown in the images below.

If you wish you can modify the incoming dialled digits using the options under the DDI/DID/TIE/MSN tab but we do not recommend this, instead catch the full number in the DDI/DID Table.

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7.7 DDI / DID Table

CO & Incoming Call > DDI / DID Table This table controls how inbound calls are routed to the various extensions and features setup on the PBX.

Any number that has been setup as the Trunk Group or a Trunk User in the Business Portal should be accounted for in this table otherwise inbound calls will not be routed correctly. The below image shows the setup for our example solution, each of the Trunk User DDI’s is directed at

the relevant extension and the Trunk Group DDI is directed to an Incoming Call Distribution (ICD) Group containing both users.

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7.8 COS Settings

System > Class of Service > COS Settings The Panasonic PBX’s use the COS settings to define what the associated features are allowed to do.

Most of the settings within these tabs are dependent on the setup and usage situation of the PBX; the values that we recommend changing within the COS Settings exist under the CO & SMDR tab. COS No. This should be the COS value that you have chosen previously

COS Name We recommend setting this as a meaningful value e.g. IPVS

Transfer to CO This should be enabled to allow calls to be transferred over the SIP

Trunk’s

Call Forward to CO This should be enabled to allow calls to be forwarded over the SIP

Trunk’s

The configuration for our example setup is shown below.

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7.9 System Options

System > System Options There are several system options that we recommend changing to make sure that the functionality of the PBX through the SIP Trunks is a full as possible.

Option 2 Tab

Under the Option 2 tab the following important values should be set; other values should be left as default:

Extension – CO Call Limitation

For Incoming Call Disable

CO – CO Call Limitation

After Conference Disable

CODEC

System CODEC A-Law

Network CODEC A-Law

The configuration for our example setup is shown below.

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Option 3 Tab

Under the Option 3 tab the following important values should be set; other values should be left as default:

Echo Cancel

Conference Disable

CO-to-CO Disable

The configuration for our example setup is shown below.

Option 4 Tab

Under the Option 4 tab the following important values should be set; other values should be left as

default:

Send CLIP of CO Caller

When call is transferred to CO (CLIP

of Held Party)

Disable

When call is forwarded to CO Disable

Send CLIP of Extension Caller

When call is forwarded to CO Enable

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The configuration for our example setup is shown below.

This concludes the System Options that we recommend changing.

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8 TROUBLESHOOTING AND USEFUL INFORMATION

8.1 Checking SIP Messaging To/From the PBX

To check the SIP messaging be sent and received by the PBX follow the below steps.

1. Within the Maintenance Console under the Utility menu select V-SIPGWXX Protocol Trace.

A processing window will appear; once it has completed and closed the trace is ready to be retrieved.

2. Again under the Utility menu select File Transfer PBX (SD Card) to PC, this will load a new page within the Maintenance Console displaying all the files available to be transferred.

3. Select the item titled PRTSIPC and click transfer, this will open a save file window allowing you

to confirm where the file will be saved on your local PC.

4. Once the file has been transferred open it using a text editor (Notepad or WordPad) to see the

SIP messaging.

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8.2 Device Fails to Register

By carrying out a SIP trace as discussed previously check that the PBX is sending SIP registration messages and whether it is receiving a response from the IPVS platform.

Check that the device has connectivity to the Internet by conducting a Ping using the Ping option under the Utility menu.

Check that the SIP ports are not being blocked by a firewall on the network. Details of the ports can be found in the Security Guide under Support Centre » Downloads » General Help & Information.

Make sure the details entered in V-SIPGWXX Account tab match those set for the Trunk Group in the Business Portal.

If you can find no problems with the configuration and setup based on the above steps please raise a ticket with Support with any SIP Protocol Traces you have taken attached.

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8.3 Outbound Calls Fail

Check that the Device is registered as above.

Check that the SIP/RTP ports are not being blocked by a firewall on the network. Details of the

ports can be found in the Security Guide under Support Centre » Downloads » General Help & Information.

Check whether any calls can be made between extensions on the PBX; if not they the issue is likely to be local to the PBX.

Check that the device is sending out SIP messaging for calls by taking a SIP Protocol Trace and looking for outbound INVITES within this.

Check that the INVITE’s are not being continually challenged by the Registrar with 401

responses; if they are then check the V-SIPGWXX Account details are the same as those set in the Business Portal.

Check that the expected outbound CLI of either the Trunk Group or the Trunk User is being presented in either the From or the P-Preferred-Identity in the outbound SIP INVITE.

If the Trunk Group or Trunk User DDI’s are being sent out in the INVITE make sure that they are in the correct format i.e. 441234567890.

If none of the above points highlight the cause please take a SIP trace of a call failing where possible and raise a ticket with Support. Include any Protocol Traces that have been taken and details of the failing call(s).

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8.4 Inbound Calls Fail

Check that the Device is registered as above.

Check that the SIP/RTP ports are not being blocked by a firewall on the network. Details of the

ports can be found in the Security Guide under Support Centre » Downloads » General Help & Information.

Check whether any calls can be made between extensions on the PBX; if not they the issue is likely to be local to the PBX.

Check that the device is receiving SIP messaging for calls by taking a SIP Protocol Trace and looking for inbound INVITES within this.

Check that the expected outbound CLI of either the Trunk Group or the Trunk User is being

presented in the TO header of the inbound SIP INVITE.

If the Trunk Group or Trunk User DDI’s are being received make sure that they match the

format of the numbers in your DDI / DID Table i.e. 441234567890.

If the DDI / DID Table is correct for the inbound INVITE make sure that the destination for the

call is available to other extensions on the PBX.

If none of the above points highlight the cause please take a SIP trace of a call failing where

possible and raise a ticket with Support. Include any Protocol Traces that have been taken and details of the failing call(s).