Guide to unlocking your Tesco Siemens DP450 IP Phone and use with VoiceHost.co.uk

Gigaset Communications have created a process that enables Tesco Internet Phone subscribers with Siemens Dual-Phone DP450 to unlock the phone from the Tesco Internet Phone service. This will enable the phone to be used with VoiceHost.
This is an approved method for unlocking the Siemens Dual-Phone DP450 defined and released by Gigaset Communications, the original supplier of the phone to Tesco.

If this procedure is not followed exactly your phone might not work any more. In this case, Gigaset Communications or VoiceHost cannot accept any warranty claims.

NOTE: The Tesco Internet IPA 1000 ATA was an IAX2 device, most VoIP providers are SIP only, including VoiceHost.

Unlock procedure
Your Phone should have the latest released Tesco software version, else this unlock procedure will not work.
On your handset:
BASE SETTINGS SOFTWARE UPDATE [OK]
→→
Select and press OK Enter the system PIN (0000) and press [OK]

The base station establishes a connection to the internet or a local PC.
[YES] Press the display key to start the firmware update. If no update is needed, this will be displayed.
To unlock the WEB interface of the DP450, you need to enter the following service code. Please use the exact code as is described below, else your phone will not work any more.

SETTINGS BASE
→
94762001
(Now "Eeprom" must be displayed)
Insert the following number sequence:
01778 06183 06191
Press OK to finish the programming.
"Saved" must be displayed on the Handset.
Then release the phone to be able to upgrade it with the latest Retail firmware version.
SETTING BASE
→
94762001
(Now "Eeprom" must be displayed)
Now insert this number sequence:
04850 04351 04864
Press [OK] to finish the programming. "Saved" must be displayed on the Handset.
After this procedure, you can upgrade your phone to use the latest retail firmware version.
SETTINGS BASE SOFTWARE UPGRADE
→→
Select and press Enter the system PIN (0000) and press [OK]
The base station establishes a connection to the internet or a local PC.
[YES] Press the display key to start the firmware update.
Please Note:
The firmware update can last up to 3 minutes.
Some people experience problems with the firmware upgrade where the process does not complete. If after 30 minutes your handset keeps flashing 'Base 1' you will need to power cycle your phone base by disconnecting the power for 10 minutes and then reconnect it.
Allow the phone to boot, we recommend leaving it a further 30 minutes to boot and upgrade. If your handset is still flashing 'Base 1' after this time, you will need to pair the handset to the base. To do this, follow the procedure below:

On your handset:
SETTINGS HANDSET REGISTER
→→
On your base:
Hold down the blue button for 30 seconds.
If all goes well, your handset should now be paired and you should be able to make a call.

Once your phone has been unlocked, you will be able to follow our standard set-up guide:

Siemens Gigaset - Wizard

The Siemens Gigaset phone can be configured to use the VoiceHost service in two ways, the first method is by using the built-in configuration wizard on the phone handset. When you connect your phone to your router and power for the first time the wizard will automatically start. To configure the phone, you will need your extension username and password. Please note, these are not the details you have to log in to the website. The required details can be located in the control panel under 'Extension Config'.

Follow the on-screen prompts, select VoiceHost and enter your extension username and password. Once you have finished answering the questions in the wizard you should be able to start making calls with VoiceHost. To test your configuration, try dialling 1234 to listen to the welcome message.

If you require more than one account/extension configured on your phone, or you are having problems accessing voicemail or IVR menus you will need to check your settings by running through the following procedures.

Web-Based Configuration

To access the web-based setup, you will need to determine the IP address of the phone. To do this you will need the handset. On the handset select 'menu > settings > base > network > IP address. The display will show you the current IP address of the phone.

Now you have obtained the IP address you can enter this address in a web browser to access the web configuration page.

Enter the default system pin of '0000' and click ok. Once logged in you will be shown a menu. Select the 'Settings' option and then 'Connections'. Select the first SIP Account 'Edit' button. You will be shown the following screen:

Please enter the following details into the fields provided:

- SIP:
Authentication Name: <extension username> See Control Panel for Details
Authentication Password: <extension password> See Control Panel for Details
Username: <extension username> See Control Panel for Details
Domain: See Control Panel for Details
Display Name: <your name>
Proxy server address: See Control Panel for Details
Proxy server port: See Control Panel for Details
Registrar server: See Control Panel for Details
Registrar server port: See Control Panel for Details

- Network:
STUN enabled: no
STUN server: See Control Panel for Details
STUN port: 5000
NAT refresh time: 20 seconds
Outbound proxy mode: Auto
Outbound proxy port: 5060

- Voice Codecs:
VoIP volume: Normal
Enable Annex B for G729: No
Selected codecs: G711 ulaw and alaw

Once the above details have been entered select the 'Set' button to save your settings.

To make sure that your 'touch tones' or DTMF will work correctly, select the 'Advanced Settings'. You will be shown the following screen:

Change the 'DTMF over VoIP Connections' to 'SIP Info' with all other options unticked. Click 'Set' to complete the setup process.

Once the phone registers, you will be able to start making calls.

Setting up the Gigaset N300IP, N510IP and GO DECT base stations

(handsets are simply paired to the base station)

Power up the N300IP, N510 or GO DECT Base station and connect a network cable into the side. Once the connection light comes on a steady blue then the device is powered up, connected to the network, and ready to use.

The base stations can have different Gigaset Model handsets paired to them, therefore, the handset model does not matter.

IP DECT Handsets for use with Gigaset Base Stations are:

• Gigaset A510 DECT Handset
• Gigaset S650H DECT Handset Pro
• Gigaset R650H DECT Handset Pro
• Gigaset CL750A GO - Sculpture DECT Handset
• Gigaset SL450A DECT Handset
• Gigaset S850A DECT Handset

Register the Handsets

1. Connect the handsets to the power outlet and let them charge.
2. When they first come on they will display the message “Please register”.
3. To start the registration process you need to press the right hand navigation key .
4. Once selected on the menu screen, use the navigation keys to move down to “Settings” and press “OK”
5. On the settings menu move down to “Handset” and select “OK”
6. On The Handset Menu select “Register H/Set” and press “OK”.
7. This will then ask for the PIN which by default is “0000”
8. The Handset will then display the Registering Procedure Screen.
9. When this screen shows you will need to press the button on the base station for a minimum of 3 seconds.
10. Once the handset is registered it may tell you that there is a message and ask you to update the firmware. This is OK to allow.

Setting up the Base Station and registering.

1. Open a web browser and go to www.gigaset-config.com. This will search for your base station and display the IP address. This will then automatically re-direct you to the base station’s web page.
2. On the login page enter the default PIN (0000) and press “OK”.
3. The first time you log in you will get a security warning. Tick the tick box and click “OK”.
4. After the security page you will be displayed with the main home page. On the Tabs at the top select the “Settings” tab.

The Settings Tab

Network > IP Configuration

If you need to set the base station with a static IP then set the IP Network details here, if it is OK to have it as DHCP then leave the settings on this page as they are. Please note that if you change to a static IP address then you will need to reload the web page.

Telephony > Connections

1. On the connections menu you need to set all of the seats that you are assigning to the base station.
2. The Seats you want to add are, by default, labelled IP1 - IP6.
3. Next to the seat you wish to configure select “Edit”
4. Once you have clicked “Edit” you will be presented with the configuration page for the Seat you want to add.
5. The first thing to do is click on “Show Advanced Settings”
6. Rename the connection to something more related to the account than IP1 .
7. Enter in the Seat details from the control panel as the username and password. The Authentication name is the same as the username.
8. Add in the name of the seat.
9. Enter in the Domain, Proxy Server and Registration Server all as per the control panel
10. Change the Registration Refresh Timer to “60”
11. Once complete click on “Set” at the bottom. This will take you back to the “Connections” page. Please note the connection will say “Registration Failed”. If you have a live internet connection simply press F5 and you should see that this is now registered. If you do have a live internet connection but after a couple of refreshes over a couple of minutes, the connection is still showing “Registration Failed”, please go back over the settings above.
12. Repeat the above step for each seat you wish to set up.

Telephony > Audio

1. On the Audio tab click on “Show Advanced Settings”
2. Under the heading for each seat that you have set up, set the audio codecs as: G711u law/G711a law/
3. Once set click on the “Set” button at the bottom of the screen.

Telephony > Number Assignment

1. Next to each handset that is registered you can name them. This name will be displayed on the handset.
2. Select the Seats that you want this to be assigned to. You must un-tick all of the other connections for each.
3. Each handset must be assigned to a different seat.
4. Once you have completed this then press “Set” at the bottom of the page.

Telephony > Advanced VoIP settings

1. Scroll to the bottom of this page and change “Use Random Ports” to “Yes” and then press “Set”.

Your Base Station and handsets should now be configured and working.

How to configure Avaya IP Office SIP Trunking with VoiceHost

Below Version 8: Avaya IP Office software older than version 8 requires a STUN server for systems that do registration.

Navigate to System in the tree-view menu on the left of the PBX GUI

LAN Settings - Enter either the public IPv4 address of the broadband circuit or one that has been issued to you the customer to use. (STUN server details can be entered here)

IP Route - An IP Route will need to be created containing the IPv4 Address of the VoiceHost Proxy, giving the route to access the gateway connected to the LAN 2 port.

SIP registrar - These settings are normally left as default

SIP Line:

• Create a new SIP Line and enter the VoiceHost FQDN as provided in your control panel.
• Set the Transport setting as shown including DNS and ITSP Proxy Address unless supplied different information.
• The SIP Credentials will need to be configured using the User Name and Password as supplied by VoiceHost and is also used for the Authentication Name and Contact.
• The SIP URI settings only need two entries as shown. One is for the outgoing calls and the other a wild card entry for the incoming DDI calls which are routed to the Incoming Call Routes by their group number.

SIP URI settings for outgoing calls

• Via is taken from the Use Network Topology Info setting on the Transport page.
• Local URI is the CLI which needs to set to line, normally the site main number and needs to be entered using the E.164 format as shown +441234567890.
• Contact, Display Name and PAI all need to be set as shown.
• Registration is selected form the drop down box and is the SIP Credentials settings.
• Incoming and Outgoing Group should be set to the same number as the SIP Line number.
• Max Calls per Channel is the amount of concurrent calls that can be made over the SIP Line. The number of channels you can set is the maximum of SIP Trunk licences the system has. Also if you have more than one SIP Line these licences must be shared between the SIP Lines.

SIP URI Setting for Incoming DDI Calls.

• Via is taken from the Use Network Topology Info setting on the Transport page.
• Local URI is set to * (star) as wild card. This will send all incoming calls to the Incoming Call Route list.
• Contact, Display Name and PAI all need to be set as shown.
• Registration is selected form the drop down box and is the SIP Credentials settings.
• Incoming and Outgoing Group should be set to the same number as the SIP Line number.
• Max Calls per Channel is the amount of concurrent calls that can be made over the SIP Line. The number of channels you can set is the maximum of SIP Trunk licences the system has. Also if you have more than one SIP Line these licences must be shared between the SIP Lines.

Incoming Call Route

• A Incoming Call Route for each DDI number needs to be created and its destination set.
• Line Group ID settings is the same as the SIP Line/Incoming and Outgoing Group number.
• Incoming Number is the DDI number entered in the E.164 format as shown +441234567890
• Destination is set normal to ring a group, user, shortcode, voicemail or voicemail action.

What are Pcap files? How do I get them? What do they mean?

A Pcap or 'Packet Capture' is also known as a log or SIP trace. As the name suggests it simply records all of the communications data and allows for it to be loaded into applications for analysis. Along with an understanding of SIP it is possible to ascertain where faults and errors occur. The location of the Pcap is important as VoiceHost can obtain them from our edge network but they can also be of value locally (from inside a local network or behind a firewall).

TLS and SRTP traffic will require decrypting and for obvious reasons we do not share encryption keys.
PCAPS also fall under various data protection laws legislation including GDPR and RIPA. You must adhere to legislation as do we.

Performing and Analysing SIP Packet Captures

Two programs are useful for this purpose but bear in mind that TLS and SRTP traffic will require decrypting and for obvious reasons we do not share encryption keys.

Collectors:

• TCPdump (Linux) www.tcpdump.org
• WinPcap (Windows and bundled with Wireshark - see below )

Analysing:

• Wireshark www.wireshark.org (Windows + MacOS)

Message Headers and Call Flows are the quickest way to determine VoIP faults

Let's describe the most important message headers in the example above.

The request start line: The string "INVITE sip:13@10.10.1.13 SIP/2.0" tells that this is an invitation to a call. It also gives the SIP address of the receiving endpoint (sip:13@10.10.1.13) and identifies the version of the protocol (SIP/2.0).

• Call-ID: This is a unique identifier of the given SIP session. It usually consists of a random string and the IP address of the sender.
• CSeq: This is an ID that identifies the particular SIP transaction. As mentioned in the previous section, the same CSeq: is always shared by a request and its related response or responses. The CSeq: identifier is composed of a sequence number (incremented for each new request) and the name of the particular request.
• From: This field ('From: "Test 15" <sip:15@10.10.1.99>;tag=as58f4201b' in our example message) contains the address of the caller with an optional display name and with optional tags. From: is a mandatory field in all SIP requests and responses. In SIP responses, From: is always a copy of the From: field in the related request message. Note that in our example, 10.10.1.99 is the IP address of the Asterisk PBX which plays the role of a SIP proxy. The actual caller (phone number 15) might be located elsewhere but the proxy will not show the actual IP address in message headers.
• To: This field contains the address of the called party. To: is a mandatory field. The To: fields in responses are copied from the related request message.
• Via: The Via: headers are used to record the route of the request. Each proxy server on the path of the message will add one Via: entry. Thanks to this, the replies can be routed back along the same path.
• Content-type: This field describes the media type of the message body. The type is usually "application/sdp", denoting the Session Description Protocol (we will look at SDP later). The message body can be sometimes empty (e.g. the REGISTER message) and then the Content-type: header is not present.
• Content-length: This is the length of the message body in octets. This header is always present but can be 0 (denoting there is no message body).

The message body carries a message of the Session Description Protocol. This message contains a description of the audio (and possibly video) channel that the calling endpoint wants to establish. We will look at the Session Description protocol later in a greater detail.
SIP Requests

SIP originally only had 6 requests (also called methods). These requests have been a part of the standard since SIP 1.0.

Individual methods can be described as:

• INVITE — This is a request to establish a call (a session). We have seen an example of the message above.
• CANCEL — This method is used to stop an INVITE that is in progress (that is, the call has not been established yet).
• ACK — The ACK request is used to confirm that the endpoint has received a final response in a transaction. Typically, after the called party accepts a call, the caller confirms the receipt of the accepting response (200 OK) with the ACK method.
• BYE — The BYE method is used to end an established call (compare with CANCEL that is used to stop the session before it has been established).
• REGISTER — The REGISTER method is used to register the SIP endpoint at the registrar server. This method does, in fact, the same as the Registration Request (RRQ) in H.323.
• OPTIONS — This request is used to ask the other party for the list of SIP methods it supports. The response may also contain the set of capabilities (i.e. audio/video codecs) of the responding party.

In addition to these six requests, several other SIP methods have been added, either in SIP 2.0 or in other individual RFCs. For example, the INFO method was defined in RFC2976. It can be used to carry application-level information that is relevant to the session, for example, participant images or account balance information. The INFO method can also be used to carry DTMF digits.

The three methods SUBSCRIBE, NOTIFY, and MESSAGE extends SIP with instant messaging features. If you send the SUBSCRIBE method, you are asking the other party to send you notifications about status changes ("available", "busy", "away", etc.). The status change notifications are then sent in the NOTIFY messages. Last, the method MESSAGE is used to send instant messages. The text of the instant message is simply transported in the body of the method (Content-Type: is usually text/plain).

Samsung OfficeSERV PBX configuration guide for the VoiceHost SIP Trunking

1. On the VoiceHost control panel disable the + for the inbound number (SIP Trunk -> Advanced - tick "No Plus")
2. Router Configuration
   The following Ports will need to forwarded to the Samsung PBX:
   · 5060 – UDP Signaling to the Processor
   · 30000 and greater – UDP, 2 ports per MGI channel
   · 40000 and greater – UDP, 4 ports per MPS channel
   · 45000 and greater – UDP, 4 ports per RTG channel

OfficeServ Configuration

1. Trunk groups –MMC603/DM4.1.2
   Create a SIP trunk group and allocate your ISP setting
2. SIP Carrier Options –MMC837/DM5.2.13
   • Registrar Address: (See your Control Panel)
   • Outbound Proxy: (See your Control Panel)
   • DNS Server: 208.67.222.222
   • Username: Username from VoiceHost (STXXXXXTXXX)
   • Auth Username: Username from VoiceHost (STXXXXXTXXX)
   • Auth Password: Password from VoiceHost
   • Alive Notify: OPTION
   • Alive Notify Time (sec): 30 seconds
   • Hold Mode: Send/Receive if you wish to use the systems MOH

All other entries can be left in default

DDI Table –MMC714/DM3.2.3 
Use the full inbound number but replace the 0 with 44

VoIP Options – MMC501/DM5.2.18
To properly support early audio on outgoing calls, e.g. BT Caller Redirect numbers (<NU TONE> “the number
called has been changed to...”), change the following setting:
Real Ringback - ON

What causes dropped telephone calls and what can I do to address the causes?

Telephone calls can be dropped for a variety of reasons but if you're suffering regularly then there are few things you can check and remedy to resolve them.

• SIP Session Timers
• SIP ALG
• Talk Off
• VAD - Voice Activity Detection
• NAT Issues

SIP Session Timers

SIP session timers tell the SIP Servers that the calls is still alive as looking for a BYE message is not reliable. If a SIP Server thinks one end has hung up (i.e stops receiving KEEP-ALIVE messages) it will drop any open call legs.

SIP ALG

Read our guidance on SIP ALG HERE

Talk Off

Talk Off happens when phone tones (DTMF) are wrongly interpreted during a call. This is easily resolved by ensuring the recommended settings for DTMF transmission OUT-OF-BAND as per RFC2833 are enabled.

VAD - Voice Activity Detection

If in doubt, turn this off as it essentially saves bandwidth by not sending full audio if nobody is talking but not everyone can support it.

NAT Issues

SIP doesn't like NAT. It's that simple. Most people have to use it as IPv4 limitations dictate and few providers fully support IPv6.

How to configure SIP Trunking on a Panasonic NCP IP PBX

Tentative Version 0.1(PSN) 18^th, July, 2013

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 changed FAX setting and NAT Keep Alive ability.

- Reject T.38 Request change to “Enable”. (Default: Disable)
  *Note SIP server does not support T.38. (Need to set reject T.38 request by PBX.)

Recommended setting
- NAT - Keep Alive Packet Sending Ability change to “Enable”. (Default: Disable)
Go to 1.Configuration - 1.Slot and select “IPCMPR Virtual Slot”. and click “Ous”.
Move mouse over “V-SIPGW16” and click “Port Property”.

Main Tab:
1. Channel Attribute:                                 Basic Channel
2. Provider Name:                                     Enter a logical name
3. SIP Server Location – Name:                 Enter your assigned server as shown in the VoiceHost control panel.
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 VoiceHost. Please note that this is just the SIP Account (user name) and DOES NOT include @FQDN For example: SIP Account (User name) = ST00000T000 Enter: ST00000T000

2. Authentication ID:                      
Enter the Authentication ID as supplied by VoiceHost. Please note that this is just the Authentication ID and DOES NOT include @FQDN For example: Authentication ID = ST00000T000 Enter: ST00000T000

3. Authentication Password:
Enter the Password as supplied by VoiceHost

Register Tab:
1. Register Ability:                                           Leave at Enable
2. Register Interval:                                        Leave at 3600
3. Un-Register Ability:                                     Leave enabled
4. Registrar Server – Name:                             Not required  * If SIP Server and Registrar Server are different, enter the Registrar Server.
5. Registrar Server – IP Address:                      Not required
6. Registrar Server port number:                      Leave at 5060

Go Back to “Slot”.
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:                    Confirm 20

Then, click“OK”. Move mouse over“V-SIPGW16” again, and click “Ins”.

Incoming Call Routing:
Go to “10. CO & Incoming call” and select “3.DDI /DID Table”
1. DDI/DID Number:                       Enter the DDI number in the format 44+PSTN Number (as below)

• Example: PSTN number=0843-9999999
• Enter: 448439999999 (Remove “0” of 0843-)

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

Outbound Call CLI:
Each extension that wishes to present an individual CLI needs to be programmed with a usable CLI. The usable CLI is a PSTN number assigned with the SIP trunk.

 i.e. if the PSTN number is 0843-9999999, the CLI to be programmed is 08439999999

Go to “Calling Party” tab.
1. From Header – User Part:         Change to PBX-CLIP

All other tabs may be left at default:
- Header Type
- From Header – SIP-URI (100 characters)
- P-Preferred-Identity Header – User Part
- P-Preferred-Identity Header – SIP-URI (100 characters)
- Number Format
- Remove Digit
- Additional Dial
- Anonymous format in “From” header
- P-Asserted-Identity header

Go to “4.Extension, 1.Wired Extension, 1.Extension Settings” & select “ISDN CLIP”

1.   Enter a valid CLI for each extension that requires it in the CLIP ID field. This setting, callee side shown ‘08439999999’.
2.   Enter the name for each extension that requires it in the Extension Name field

This setting, what characters shown callee side is now testing.

[T.38 Tab]

1. Reject T.38 Request from Network:      Change to Enable

All other tabs may be left at default
- T38 FAX Max Datagram
- T38 FAX UDPTL Error Correction - Redundancy
- T38 FAX UDPTL Redundancy count for T.30 messages
- T38 FAX UDPTL Redundancy count for data

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

DOWNLOAD HERE