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.

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

A browser extension which converts telephone numbers into clickable links to call using the VoiceHost desktop softphone application

How does it work?

This extension recognizes phone numbers on web pages and converts them into clickable links.

This is done by passing the phone number to the configured system protocol handler and from there to the application which registered this protocol handler. Just like an URL starting with “http”, a link can also start with other protocol specifiers, e.g. “tel”, “sip” or “callto”.

Phone numbers on a web page recognized by the extension and highlighted with an optional icon. When you click on a number. By clicking on this menu item the phone number is passed to the dial-pad of the desktop application “as is”.

Download the Extension for your browser:

Outbound calls error with "all circuits busy" or "congestion":

This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration (If Asterisk, swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly.

Outbound calls fail with SIP error 488 (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):

Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm
If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. 8000/20i - 8000Hz at 20ms) cannot interwork with 16000/30i - 16000Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to use one of the above codecs.

Inbound calls fail with SIP error 408 (Request Timeout):

Check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus, e.g. if Asterisk is configured to use plus somewhere else. Check the trunk is registered. Ascertain how long the 408 error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly.

Calls fail with SIP error 503, I-SUP errors 34 or 38:

If our platform replies back with 503 it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.