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 HELIATEL platform to be able to successfully make calls.
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.
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 VSIPGW 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. 
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 HELIATEL platform.
Click OK once you have finished entering the necessary settings.
The below image shows the example settings for our solution. 
Port Property
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.
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. 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 |
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. 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 |
The configuration for our example setup is shown below.
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) | 240 |
| Un-Register Ability when port INS | Enable |
| Registrar Server - Name | Leave Blank |
| Registrar Server – IP Address | Leave Blank |
| 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:
| Session Timer Ability | Enable(Passive) |
| Session Expire Timer (s) | 900 |
| Refresh Method | re-INVITE |
| Proxy-Require Option | Leave Blank |
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 | 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 |
The configuration for our example setup is shown below.
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 |
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.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 |
The configuration for our example setup is shown below.
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 | ToS |
| RTP QoS-DSCP | Leave blank |
| RTCP Packet Sending Ability | Enable |
| RTCP Packet Interval | 5s |
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.
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.
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.
C:\Users\Dar\Desktop\More Stuff\panasonic\screenshots\Panasonic-Trunk-Group-TRG-Settings-Main.png 
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:
- 10 Digit Local Calls
- 11 Digit Long Distance Calls
- 911 Emergency Calls
The below image taken from our solution shows some example rules from our proposed solution, it does not cover all dialling options.
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 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 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.
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.
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.
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.
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 as 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 | u-Law |
| Network CODEC | u-Law |
The configuration for our example setup is shown below.
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 |
The configuration for our example setup is shown below.
This concludes the System Options that we recommend changing.
TROUBLESHOOTING AND USEFUL INFORMATION
Checking SIP Messaging To/From the PBX
To check the SIP messaging be sent and received by the PBX follow the below steps.
- 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.
- 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.
- 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.
- Once the file has been transferred open it using a text editor (Notepad or WordPad) to see the SIP messaging.
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 HELIATEL 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.
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).
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. 14036687777.
- 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).
