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SIP Configuration Notes – HELIATel Premium
Use the specific configuration guide below as an example to configure an SV8100 or SL11100/SL1000 PBX for connection to the service described above via SIP trunks.
Recommended Software Versions
| SV8100: CCPU | vC9.00 |  | SL1X00: CCPU | v4.0 |  |
System Programming
The following items should be changed – all other items are considered irrelevant and as such left as default. Screenshots are for example purposes only and will have been taken from the PBX under test but will apply to the other PBXs listed on the cover of the certificate. Any differences in programming will be documented where necessary.
CCPU IP Address
| Easy Edit | PRG | Item | Setting |
|---|---|---|---|
| Advanced Items > IP/SIP > Blades > CCPU IP Address | 10-12-09 | IPL/VOIPDB IP Address | Set according to customers network requirements |
| 10-12-10 | IPL/VOIPDB Subnet Mask | Set according to customers network requirements | |
| 10-12-03 | Default Gateway | Set according to customers network requirements | |
| 10-12-02 | Default Gateway Subnet Mask | Set according to customers network requirements | |
| 10-12-01 | IP Address | Must be in a different network range to IPL IP Address (10-12-09) |
VoIP Resource IP Address
| Easy Edit | PRG | Item | Setting |
|---|---|---|---|
| Advanced Items > IP/SIP > Blades >VoIP Resource IP Address | 84-26-01 | VoIP Gateway IP Address | Set according to customers network requirements and IPL channel capacity IPLA: 32 Channel = 2 x VoIP Gateway IP addresses 64 Channel = 4 x VoIP Gateway IP addresses 128 Channel = 8 x VoIP Gateway IP addresses IPLB: All Channels = 1 x VoIP Gateway IP address |
SIP Trunk Assignment
| Easy Edit | PRG | Item | Setting |
|---|---|---|---|
| Advanced Items > IP/SIP > SIP Trunks > SIP Trunk Assignment | 10-40-01 | IP Trunk Availability | Set to Enabled |
| 10-40-02 | IP trunk Port Count | Set to number of SIP trunks required. Associated IP Trunk Licenses must be installed on PBX. |
VoIP Trunk Type
| Easy Edit | PRG | Item | Setting |
|---|---|---|---|
| Advanced Items > IP/SIP > SIP Trunks > VoIP Trunk Type | 10-03-02 | Trunk Type | Set VOIPU card trunk ports to SIP |
Networking Mode
| Easy Edit | PRG | Item | Setting |
|---|---|---|---|
| Advanced Items > IP/SIP > Blades > Networking Mode > Networking Mode | 10-23-01 | System Interconnection | Set to Enabled |
| 10-23-02 | IP Address | Set to S52.117.200.68 | |
| 10-23-04 | Dial Number | The first digit(s) that will be dialled | |
| 10-28-01 | Domain Name | Set according to customers network requirements | |
| 10-28-02 | Host Name | Set according to customers network requirements | |
| 10-28-04 | UserID | User ID, as supplied by HELIATel Premium | |
| 10-29-14 | Carrier Choice | Carrier Choice set as Carrier B | |
| 10-29-16 | Register Sub Mode | Set to Disabled | |
| 84-13-28 | Audio Capability | G.711_PT | |
| 84-13-32 | DTMF Relay Mode | Set to RFC2833 |
Notes
- DDIs can be configured if required, using the same procedure as for ISDN trunks
- SIP calls are sent “en bloc”. This means that the External Call Inter digit timer (PRG21-01-03) must expire before the call is set up. This can be reduced, but will have an impact on ISDN trunks also. The user can dial # to indicate “end of dialling” instead if required.
- CLI can be sent out using either 21-17 IP Trunk Calling Party Number Setup for Trunks or 21-19 IP Trunk Calling Party Number Setup for Extensions. If nothing is defined against the extension or trunk, or CLI is restricted through further programming, then no outbound CLI is sent.
Network Configuration
If Public IP addresses are assigned to the SV8100 IPLA and VoIP Gateways, then there should be no
network configuration required.
If there is one public IP address assigned, and NAT is used, it is necessary to configure Port Forwarding
on the router:
- Port 5060 should be forwarded to the IPLA IP address
- Port 10020 – 10275 should be forwarded to the VoIP Gateway IP addresses
- NAPT should be enabled in PRG10-12-06 and the Public IP address should be entered into
PRG10-12-07.
Known Limitations/Comments
- The service uses “Networking Mode” which means that the system does not maintain a
registration. This means that the system is unable to determine if the SIP server is available, and
if it is acceptable to set up calls. - The service only supports codec G.711.
- At the time of publishing the service does not support T38 fax mode. Should the carrier add this to the service it will not be added to the certificate until further testing has been completed.
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INTRODUCTION
This document describes the configuration procedures required for the Panasonic KX-NCP and KXTDE 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.
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 HELIATEL platform in one or more of the following headers of the SIP message transmitted by the IP-PBX (listed in order of significance):
- Diversion
- P-Asserted-Identity
- Remote-Party-Id
- From
If you see a 604 SIP message back from IPVS then the device is not presenting the correct CLI to the HELIATEL platform or the account is not present on the HELIATEL platform.
SIP Trunking DDI Users must have an active Public Number (DDI) on the HELIATEL 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 HELIATEL 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
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.
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: 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 HELIATEL platform to be able to successfully make calls. 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. 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. 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 VSIPGW card out of service. Once all steps have been carried out the card should be brought back in to service. Select the Shelf Property link for the V-SIPGW card. Under the Main tab the following important values should be set: The below image shows the example settings for our solution. The default values under this tab do not need to be changed. Click OK once you have finished entering the necessary settings. 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 HELIATEL platform. Click OK once you have finished entering the necessary settings. The below image shows the example settings for our solution. Select the Port Property link for the V-SIPGW card. 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. Under the Main tab the following important values should be set for your desired primary channel: Part of the configuration for our example setup is shown below. Under the Account tab the following important values should be set for your desired primary channel: The configuration for our example setup is shown below. Under the Register tab the following important values should be set for your desired primary channel: The configuration for our example setup is shown below. No values should be entered under this tab as STUN is not used. Under the Option tab the following important values should be set for your desired primary channel: Under the Calling Party tab the following important values should be set for your desired primary channel: The configuration for our example setup is shown below. Under the Called Party tab the following important values should be set for your desired primary channel: 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: The configuration for our example setup is shown below. 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: If you wish to fax through the PBX using T.38 then we would advise leaving the values in this tab as the defaults. 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. We would advise leaving these values as the defaults unless advised otherwise. Under the Supplementary Service tab we recommend setting the following values for all used channels: The configuration for our example setup is shown below. 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. Group > Trunk Group > TRG Settings 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. The configuration for our example setup is shown below. 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: The below image taken from our solution shows some example rules from our proposed solution, it does not cover all dialling options. 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 HELIATEL platform). This should be in the format of the DDI set in the Business Portal for the Trunk Group/User i.e. 14036687777. 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. 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. The configuration for our example setup is shown below. 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. 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. 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. The configuration for our example setup is shown below. 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 as full as possible. Under the Option 2 tab the following important values should be set; other values should be left as default: The configuration for our example setup is shown below. Under the Option 3 tab the following important values should be set; other values should be left as default: The configuration for our example setup is shown below. Under the Option 4 tab the following important values should be set; other values should be left as default: The configuration for our example setup is shown below. This concludes the System Options that we recommend changing. To check the SIP messaging be sent and received by the PBX follow the below steps.
Panasonic KX-NCP or Panasonic KX-TDE
(Dependant on your IP-PBX type)PANASONIC IP-PBX CONFIGURATION
SIP Settings
Shelf Property
Main Tab
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 
Timer Tab
Card Property
Port Property
Main Tab
Channel Attribute Basic Channel Provider Name We recommend setting this as a meaningful value e.g. HELIATEL Premium SIP Server Location - Name sbc2.heliatel.ca SIP Server Location – IP Address This is only required if the PBX cannot access a DNS server, do not enter unless necessary; 52.117.200.68 SIP Server Port Number 5060 SIP Service Domain sbc2.heliatel.ca Subscriber Number Trunk Group DDI as provisioned in the Business Portal e.g. 14036687777 
Account Tab
Username Trunk Group DDI as provisioned in the Business Portal e.g. 14036687777 Authentication ID Trunk Group Username as provisioned in the Business Portal e.g. 14036687777 Authentication Password Trunk Group Password as provisioned in the Business Portal 
Register Tab
Register Ability Enable Register Sending Interval (s) 240 Un-Register Ability when port INS Enable Registrar Server - Name Leave Blank Registrar Server – IP Address Leave Blank Registrar Server Port Number 5060 
NAT Tab
Option Tab
Session Timer Ability Enable(Passive) Session Expire Timer (s) 900 Refresh Method re-INVITE Proxy-Require Option Leave Blank Calling Party Tab
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 PBX-CLIP This should match what you have entered in the From Header – User Part field. P-Preferred-Identity Header – SIPURI Leave Blank Number Format National Remove Digit 0 Additional Dial Leave Blank Anonymous format in "From" header Display name and SIP-URI P-Asserted-Identity header Disable 
Called Party Tab
Number Format National Type To header Voice/FAX Tab
IP Codec Priority The 3 priority boxes allow you to specify the preferred codec order for the PBX, in general we recommend;
1st – G.711Mu
2nd – G.729A
3rd – None
Packet Sampling Times (s) These values should be set as follows for each codec;
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 
RTP/RTCP Tab
RTP QoS Ability ToS RTP QoS-DSCP Leave blank RTCP Packet Sending Ability Enable RTCP Packet Interval 5s T.38 Tab
T.38 Option Tab
DSP Tab
Supplementary Service Tab
CLIR Yes CNIP (Send) Yes CNIP (Receive) No 
Assigning Additional Channels
Trunk Group Settings
Main Tab
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 
Dialing Plan
Extension Settings
CO Line Settings
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. 
DIL Table & Port Settings
DDI / DID Table
COS Settings
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 
System Options
Option 2 Tab
Extension – CO Call Limitation For Incoming Call Disable CO – CO Call Limitation After Conference Disable CODEC System CODEC u-Law Network CODEC u-Law 
Option 3 Tab
Echo Cancel Conference Disable CO-to-CO Disable 
Option 4 Tab
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 
TROUBLESHOOTING AND USEFUL INFORMATION
Checking SIP Messaging To/From the PBX
Device Fails to Register
Outbound Calls Fail
Inbound Calls Fail
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Configuring a Generic SIP trunk on 3CX Phone System is simple and easy. There is a few items that are required including:
- Registar Server Address
- SIP Username
- SIP Password>
- Concurrent Calls
- Primary Phone Number
I'll show you how to configure a SIP trunk in 3CX using the information we provide at HeliaTel.
Configure your "Outbound Rule"
In order to place phone calls out of 3CX Phone System, an outbound rule(s) must be created. We recommend at least 3 outbound routes are create. These routes are:
- Standard 10 digit local numbers (a prepend a 1)
- Standard 11 digit long distance
- 911 Emergency phone number
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