Updated/extended functions
u Changes to getting the phone started.
For example, Gigaset.net assistant is only started when you first open the Gigaset.net directory. Page 5 Since several consecutive RTP ports are required for each VoIP connection, you can now specify a port number range for the RTP ports when configuring the telephone. Page 23 The signalling of calls made to a number that is not assigned to a handset as a receive number has changed. Page 25 If you do not update the firmware or VoIP provider data when prompted, the phone will remind you again at a later date. (Only for devices manufactured after May 2009) Page 26 When defining dialling rules, you can use the new option Use Area Codes to specify whether or not the "automatic area code" is also to be dialled. Page 27 The key combination for checking the MAC address of the base has changed Page 27
New information on troubleshooting/problem analysis
u New functions (e.g. immediate download of a provider profile) have been added to the
service information that you can use during an external call (e.g. with the Gigaset service). Page 28 u If your phone is connected to a NAT router, the NAT can cause problems during VoIP telephony (especially if you connect multiple VoIP telephones to your router). Notes on resolving these problems can be found in these amendments. Page 29 u The table of VoIP status codes that you can display on the screen has been extended. The extended table can be found in these amendments. Page 31
Function no longer in use
u When dialling, you can no longer select the line type by adding # or *.
(Only for devices manufactured after May 2009) Page 34
Description of new and updated functions Changes to phone setup process
Description of new and updated functions
Changes to phone setup process
This section amends the section "First steps Making settings for VoIP telephony" in the user guide for your Gigaset VoIP phone.
The procedures for "Making settings for VoIP telephony" have changed as follows. 1. Auto-configuration: After you have started the installation assistant and entered the system PIN, the following is displayed:
Do you have a code for auto configuration?
If you have received an auto-configuration code (Activation Code) from your VoIP provider: Press the key below Yes on the display screen. You are prompted to enter the code. Use the keypad to enter the auto-configuration code (max. 32 characters) and press OK. All data necessary for VoIP telephony is loaded directly from the Internet to your phone. The handset returns to idle status. The configuration is complete. If your VoIP provider has supplied you with an authentication name/password and, where applicable, a user name: Press the key below No on the display screen. The VoIP configuration is then performed as described in the user guide for your telephone.

2. Gigaset.net assistant: After you have completed the VoIP configuration, i.e. after entering your user data or the auto-configuration code for your VoIP account, the Gigaset.net assistant is no longer started (the step "Entering your name in the Gigaset.net directory" in the telephone user guide can be skipped). After you have entered your user data or the auto-configuration code, the handset reverts to idle status. The Gigaset.net assistant is started when you open the Gigaset.net directory for the first time ( Page 7). You can then enter your name in the Gigaset.net directory. 3. If a firmware update for your telephone is available in the Internet: In this case, the message New firmware available is displayed if you press the flashing message key after starting your handset and connecting the base. Perform the firmware update (press the right display key Yes). Once the update has been completed (after approx. 3 minutes) the handset's idle display appears again and the message key f flashes. If you press f, the following is displayed: Start wizard for entry of VoIP connection data?. You can then start the connection assistant as described in the user guide.
Description of new and updated functions Starting the Gigaset.net assistant when first opening the Gigaset.net-directory
Starting the Gigaset.net assistant when first opening the Gigaset.netdirectory
After setup, you can use the Gigaset.net assistant to enter your name in the Gigaset.net directory, i.e. create a Gigaset.net directory entry for your telephone. To do so, open the Gigaset.net directory. Precondition: Your handset is in idle status. s Press and hold. q If necessary, select Gigaset.net from the list of available online directories and press OK. The following appears in the handset's display:
Gigaset.net Start assistant for Gigaset.net?
Press the display key Yes to start the assistant.
Please note The Gigaset.net assistant is only opened the first time you open the Gigaset.net directory. If you press No, the assistant will be cancelled and not restarted. You can then enter your name using the Gigaset.net directory menu (Options Own details).
Gigaset.net Your Gigaset.net nickname:
Using the keypad, enter the name that you would like to be listed under in the Press the right-hand display key OK.
Version 4, 29.10.2007 Gigaset.net directory. The name may contain a maximum of 25 characters. If there is already an entry with this name, you are requested to enter a different name. If an entry was created successfully in the Gigaset.net directory, the message "Your user name has been added to Gigaset.net!" is displayed. The handset returns to idle status. 7

Check your Internet connection and repeat the request at a later time.
Registration for personalised info services
If a special registration with user name and password is required to access an info service, the following is displayed after the service has been called up (example):
Login Username: Password:

Username Password

Enter the user name that you have agreed with the info service provider. Enter the password associated with this user name. Press the display key to send the registration data.
If registration was successful, the requested info service is displayed.
If registration failed, a message to this effect appears on the display loading requested information, Page 9. Please note Please remember that registration data is case sensitive.

Messages when

Operating the Info Center
Depending on the type of info service requested, you can carry out the following actions:
Scrolling within an info service
You can use s to scroll downwards within an info service, and t to scroll up (back).
Skipping back to the previous page
Press the left display key.
Skipping back to the Info Center menu
You want to go offline: Press and hold the end call key a, the handset returns to idle status.
Briefly press the end call key a.

Selecting a hyperlink

u Hyperlink to further information:
If the page contains a hyperlink to further information, this is indicated by the g icon. If a page is opened using hyperlinks, the first hyperlink is highlighted. Using the control keys (s and/or t) as required, you can navigate to the hyperlink that you would like to select. The hyperlink is then highlighted by bars. Press the right display key Link, to open the relevant page. u Hyperlink to a phone number: If a hyperlink contains a phone number, you can copy the number to the local directory or call the number directly (Click-2-Call functionality). Select the hyperlink using the s and/or t keys, as required. You can identify a hyperlink of this type by the fact that Call is shown above the right display key. Press CopyToDir if you want to copy the phone number to your handset's local directory. Or: Press Call to call the stored number.

u If you have selected a VoIP connection and the attempt to transmit the SMS messages

Please note

fails, the SMS with error status is stored in the incoming message list. Even if you have activated your fixed line network connection as an alternative connection ( Page 20), the telephone does not attempt to send SMS messages via the fixed line network. u If you have selected a VoIP connection as a send line and this is deleted from the configuration, the first VoIP connection in the configuration will be used.
Note on writing, sending and receiving SMS messages etc.
Regardless of your send line settings (fixed line network or VoIP) you can write, send and receive SMS messages as well as request SMS notifications as described in the user guide for your phone ( chapter "SMS (text messages)"). If your VoIP provider supports the relevant features, you can also use personal mailboxes, send SMS messages to e-mail addresses, or request SMS info services. Please note Every SMS addressed to one of your numbers (VoIP or fixed line network) is displayed on all registered handsets with SMS functionality, even if the phone number addressed is not assigned to the handset as a receive number.
Description of new and updated functions Reading e-mail messages on the handset
Reading e-mail messages on the handset
This section amends the chapter "E-mail notifications" in the user guide for your Gigaset VoIP phone.
Your phone will notify you when new e-mail messages have been received on your incoming e-mail server. Using the handset, you can now display the sender, date/time of receipt, subject and message text for each e-mail in the inbox.

Preconditions:

u You have set up an e-mail account with an ISP. u The incoming e-mail server uses the POP3 protocol. u You have saved the name of the incoming e-mail server and your personal e-mail access
data (account name, password) in the phone ( user guide for your phone, Web configurator page: Settings Messaging E-Mail).
v Messaging E-mail Or if new e-mail messages have been received (the message key f flashes): f E-mail: The telephone establishes a connection to the incoming e-mail server. A list (inbox) of e-mail messages that are stored there will be displayed. The sequence in which the e-mail messages are displayed is dependent on your POP3 server. Generally speaking, the new unread messages appear before old messages that have been read.

Opening the inbox

Opening and reading e-mail messages
Select e-mail entry. Press the display key.
The subject (Subject:) and text (Text:) of the e-mail message are displayed. Any attachments to the e-mail are not displayed. Example display:
E-mail Viewer Subject: Invitation Text: Hello Anna, are you coming to the football match on Friday? 1 2

Options

1 Subject of the e-mail message. A maximum of 120 characters are displayed. 2 Text of the e-mail message (abbreviated if necessary). A maximum of the first 640 characters of the subject and message text are displayed in total (Subject + Text + "Subject:" + "Text:" = 640 characters).
Press the display key to return to the inbox.
u If the e-mail message contains more than just unstructured text, a brief message to this
effect is displayed. The Subject of the message is then displayed. ently to how they appear on the PC e-mail client.
u If the subject and/or message text are in HTML format, they may be displayed differ-
Viewing e-mail sender's address
Precondition: You have opened the e-mail message to read ( Page 14). Options Press the display key. From Select and press OK. The sender's e-mail address is displayed in full (if necessary over several lines). Press the display key to return to the inbox. Example:
From firstname.surname@ mailprov.com
Deleting the e-mail message
You have opened the inbox: q Select e-mail entry. Delete Press the display key. Or: You have opened the e-mail message to read ( Page 14) or request the e-mail sender's address to be displayed ( Page 15): Options Press the display key. Delete E-mail Select and press OK. The e-mail is deleted from the incoming e-mail server.
Description of new and updated functions Deactivating your handset's microphone
Deactivating your handset's microphone
As well as muting the handset as described in the user guide (press left on the u control key, the other party hears hold music), you can deactivate your handset's microphone during an external call. The other party cannot hear you, but you can still hear them. You can also deactivate the microphone during a conference call or when call swapping.
Turning off the microphone
Press the display key to deactivate the handset. Your handset's microphone is deactivated. The display shows Microphone is off. Press the display key to switch the microphone back on. The other party can hear you again.

Switching the microphone back on
Please note The microphone is automatically switched on again in the following scenarios: u If, during an external call (you have switched the microphone off), you establish a second connection, either by accepting a waiting call or by successfully connecting to an external/internal consultation call, the microphone is turned on. If you go back to the first party, the microphone remains switched on. (If you reject a waiting call or are unable to connect to a consultation call, the microphone remains switched off.) u If you deactivated the microphone while call swapping, it is reactivated for both connections, as soon as they are connected to the other caller. u If you have deactivated the microphone during a conference call, the microphone is reactivated when you terminate the conference call by selecting Options End Conference (call swapping).
Description of new and updated functions Network services during an external call
Network services during an external call
This section amends the sections "Network services Further network services in the fixed line network" and "Network services Further network services for VoIP" in the user guide for your Gigaset VoIP telephone.
Some network services that were previously accessed via display keys are now provided via the context menu. To open the pop-up menu you must press the display key Options.
u Fixed line network: You have requested the following network services from your fixed

line network provider.

u VoIP: Your phone permits two parallel VoIP connections.
( user guide for your phone, Web configurator Settings Telephony Audio). The following functions are affected u Consultation call During an external call via VoIP or the fixed line network: Press the display key Options. Select External Call and press OK. Enter a number or copy it from the directory and press OK. The first party is placed on hold and hears hold music. u Accepting a waiting call Precondition: Call waiting is activated ( user guide for your phone). You are conducting an external call via VoIP or the fixed line network. A second caller (waiting call) is signalled: Press the display key Options. Select Accept waiting call and press OK. The first party is placed on hold and hears hold music. u Initiating a conference call You are call swapping and want to talk to both parties simultaneously: Press the display key Options. Select Conference and press OK. u Ending a conference call (call swapping) Press the display key Options. Select End Conference and press OK.

Open the following Web page: Settings IP Configuration.

Area: HTTP proxy

Enable proxy Click the Yes option if your phone is to handle HTTP calls via your network's HTTP proxy server. If you select No, the phone will attempt to access the Internet directly. Proxy server address Enter the URL of the proxy server to which your phone is to send HTTP calls. The proxy server then creates the connection to the Internet. Proxy server port Enter the communication port used on the HTTP proxy server (number between 0 and 55000). It is mainly port 80 that is used.
Now select Set to save your settings.
Description of new and updated functions Activating/deactivating the STUN server of the Gigaset.net connection
Activating/deactivating the STUN server of the Gigaset.net connection
This section amends the chapter "Web configurator Configuring the Gigaset.net connection" in the user guide for your Gigaset VoIP phone.
The Gigaset.net connection is preconfigured in your phone. The Gigaset.net uses a STUN server as standard. In the sent data packets, Gigaset.net replaces the private IP address of your phone with its public IP address. If you operate your phone with a symmetrical NAT router, STUN cannot be used. Otherwise, when making Gigaset.net calls you will not be able to hear the caller. In this case, deactivate STUN for the Gigaset.net connection.
Open the following Web page: Settings Telephony Connections. Select Edit in the Gigaset.net area. Select Set to save the changes.
STUN enabled Select No to deactivate STUN. Select Yes if you want your phone to use STUN.
Activating the fixed line network connection as an alternative connection
You can activate the fixed line network connection on your phone as an alternative connection. If an attempt to establish a connection via VoIP then fails, an attempt is made automatically to establish the connection via the fixed line network. An alternative connection would be used in the following cases: u your VoIP connections are busy u the SIP server for the VoIP connection cannot be accessed u the dialled VoIP connection has not yet been configured or has not been configured correctly (e.g. incorrect password) u the base does not have a connection to the Internet, e.g. because your router is deactivated or not connected to the Internet.
u SMS messages that are to be sent via a VoIP connection are not sent via the fixed line

Exceptions

network connection as an alternative. The SMS message is stored in the incoming message list with an error status. Your handset's message key will flash. u If you enter a VoIP line suffix (#1 to #6) or press the IP display key before dialling, the connection is not established over the fixed line network as an alternative. u If you dial a URI or IP address instead of a phone number, the connection cannot be established via the fixed line network. Version 4, 29.10.2007

Open the following Web page: Settings Telephony Number Assignment.
Description of new and updated functions R key function for VoIP Hook flash/call diversion

Area Default Connection

If you want to activate the fixed line network connection as an alternative connection,
click the Yes option next to Automatic Fallback to Fixed Line. Select No to deactivate the function. Now select Set to activate your settings.
R key function for VoIP Hook flash/call diversion
This section replaces/amends the sections "Web configurator R key function for VoIP (Hook flash)" in the user guide for your Gigaset VoIP phone.
Using your phone's Web configurator, you can assign a special feature of your VoIP provider to the S key. Alternatively, you can use the S key for call diversion (call transfer).
Assigning the signal for a provider feature to the S key
To be able to use a special feature of your VoIP provider, your phone must send a specific signal (data packet) to the SIP server. You can assign this "signal" to your phone's R key. If you press the R key during a VoIP call the signal will be sent to the server.

Precondition:

u DTMF reminders via SIP info messages is activated, i.e. the SIP Info option on this web u The S key is not used for call transfer, i.e. Use the R key to initiate call transfer with the SIP

page is activated.

Open the following Web page: Settings Telephony Advanced Settings.

Area Hook Flash (R-key)

Refer method. = No is set for call transfer ( Page 22). If one of these preconditions is not fulfilled, the field in the Hook Flash (R-key) area is hidden.
In the Application Type (maximum 31 characters) and Application Signal fields (maximum Now select Set to save your settings.
15 characters), enter the data that you have received from your VoIP provider. The setting for the S key applies to all registered handsets.
Configuring the S key for call diversion (call transfer)

If you are transferring calls via VoIP connections, you can connect the two external callers (provider-dependent). You can configure settings for this type of call transfer.

Area Call Transfer

Use the R key to initiate call transfer with the SIP Refer method. If you select Yes, the external parties you are toggling between will be connected when you press the R key S. Your connections with the parties will be terminated.
u You can also activate the Transfer Call by On-Hook option. In this case, the two external
parties are connected with one another when you press the end call key a. To do so, you must use the Web configurator to define the preferred protocol to be used for call diversion ( user guide for your phone). u If you have deactivated both options, i.e. both Use the R key to initiate call transfer with the SIP Refer method. and Transfer Call by On-Hook, you can also divert a VoIP call using Options Call Transfer.
Description of new and updated functions Defining local communication ports for VoIP
Defining local communication ports for VoIP
This section replaces the section "Web configurator Defining local communication ports for VoIP" in the user guide for your Gigaset VoIP phone.
Specify which local communication ports (port numbers) the telephone is to use for VoIP telephony. The ports must not be used by any other subscriber in the LAN. The following communication ports are used for VoIP telephony: u SIP port Communication port via which the phone receives (SIP) signalling data u RTP port RTP ports are used to receive voice and control data. Three consecutive, even port numbers are required for each VoIP connection. You define a fixed number for the SIP port and a fixed number range for the RTP port or set your phone so that it uses free ports from a specified port number range ( Use random ports).
Area Listen ports for VoIP connections
Use random ports Click No if you want the phone to use the ports specified in the SIP port and RTP port fields. Select Yes, if you do not want the phone to use fixed ports for SIP port and RTP port, but rather to use any free ports from predefined ranges of port numbers. The use of random ports makes sense if you want several phones to be operated on the same router with NAT. The phones must then use different ports so that the router's NAT is only able to divert incoming calls and voice data to one (the intended) phone.

Use random ports = No

SIP port Specify the port number for the SIP port. Enter a number between 1024 and 49152 in the field. The default port number for SIP signalling is 5060. The port number specified must not be in the RTP port number range. RTP port Specify a range of port numbers that are to be used as RTP ports. This range must be used in the LAN (router) for the phone. Enter the lowest port number in the left-hand field and the highest number in the right-hand field (numbers between 1024 and 55000).
Size of the port number range: The difference between the port numbers must be at least 6 if you permit two simultaneous VoIP calls on your phone. It must be at least 4 if you only permit one VoIP call ( user guide for your phone, Web configurator Settings Telephony Audio). The lower of the port numbers in the range (in the left-hand field) must be an even number. If you enter an odd number, the next lowest even number will be selected automatically (e.g. if you enter 5003, then 5002 is set automatically). The default port number for voice transmission is 5004.

Use random ports = Yes

SIP port Enter the port number range from which the SIP port is to be dialled. Enter the lowest port number in the port number range in the left-hand field and the highest number in the right-hand field (numbers between 1024 and 49152). This port number range must not overlap the range specified for RTP port. The default range is 5060 to 5076. RTP port Specify a range of port numbers from which the RTP ports are to be dialled. Enter the lowest port number in the port number range in the left-hand field and the highest number in the right-hand field. The default range is 5004 to 5020.
Description of new and updated functions Amendment to "Call signalling and number assignment"
Amendment to "Call signalling and number assignment"
The section amends the sections "Accepting calls", "Web configurator Assigning send and receive numbers to handsets" and "Web configurator Assigning receive numbers to the answering machine" in the user guide for your Gigaset VoIP phone.
Signalling incoming calls
If you have not assigned any receive numbers, either to the answering machine or the registered handsets, calls to all connections will be signalled on all handsets. If you have assigned receive numbers, your handset will only indicate calls to receiving numbers assigned to this handset. Please note the following cases: u If the phone number is not assigned to a handset or an answering machine as a receive number, all calls to this number are signalled on all handsets. u If the phone number is not assigned to a handset, but is assigned to the answering machine, the call is not signalled on any handset and is accepted by the answering machine. u Calls to your phone's IP address are signalled on all handsets.
Amendment to "Searching in the online directory"

Description of new and updated functions Amendment to "Updating VoIP provider settings"
Amendment to "Updating VoIP provider settings"
The "Profile update reminder" described here only applies to devices for which a firmware version more recent than 02.140 was already loaded when the device was bought (manufactured after May 2009).
After the first download of the VoIP provider settings, your phone will check daily whether a newer version of these settings is available on the Internet. If this is the case, the message New profile available is displayed when the handset is in idle status and the message key f flashes. If you do not want to perform an update at this time, press the message key f and respond to the subsequent prompt with No. The handset switches to idle status. Your phone will then remind you at a later date about the new profile (New profile available will be displayed again).
Amendment to "Defining dialling plans"
This section amends the section "Web configurator Defining dialling plans Cost control" in the user guide for your Gigaset VoIP phone.
When defining dialling plans, the additional option Use Area Codes is now available ( user guide for your phone, Web configurator page Settings Telephony Dialling Plans). Activate the option Use Area Codes for all VoIP calls, if the automatic area code is to precede all phone number(s) for which the the dialling plan is defined. You can define the automatic area code on the Web page Settings Telephony Dialling Plans under Area Codes. Please note In the case of dialling plans for emergency numbers, you should always deactivate the option Use Area Codes.
Correction to "Checking the base MAC address"
This section replaces the section "Base settings Automatic firmware update" in the user guide for your Gigaset VoIP phone.
To display the base MAC address, press the following keys in sequence while the handset is in idle status. On the Gigaset S67H or S68H handset: v N5OM5 On the Gigaset C47H handset: v 55OM5
Press and hold the end call key a to return to idle status.
Description of new and updated functions Checking extended service information for the base

Edited/extended table of VoIP status codes
This table replaces the table of VoIP status codes provided in the appendix of the user guide for your telephone.
In the following tables you will find the meaning of the most important status codes and messages.
Status Meaning code IP configuration error: IP domain not entered. IP configuration error: SIP user name (Authentication Name) not entered. This is shown, for example, when dialling with a line suffix, if no connection is configured for the suffix on the base. IP configuration error: SIP password (Authentication password) not entered. The called party can be reached under several phone numbers. If the VoIP provider supports this, a list of the phone numbers is transmitted as well as the status code. The caller can select to which number he wants to make the connection. Permanently redirected. The called party can no longer be reached under this number. The new number is transferred to the phone together with the status code, and the phone then no longer accesses the old number but dials the new address immediately. Temporarily redirected. The phone is informed that the called party cannot be reached under the dialled number. The call is redirected for a limited period. The phone is also notified of the length of the redirection. The query is sent to a different "proxy server", e.g. to balance incoming queries. The phone will once again make the same query to another proxy server. This is not a redirection of the address per se. Other service: The query or call could not be transferred. But the phone is notified what other options there are to be able to connect the call. Wrong call Not authorised The requested service is not supported by the VoIP provider. Wrong phone number. No connection on this number. Example: In a local call you have not dialled the area code although your VoIP provider does not support local calls. Method not permitted. Not acceptable. The requested service cannot be provided. Proxy authentication required. The party cannot be reached (e.g. account has been deleted).

34 300

403 404

405 406

407 408
Description of new and updated functions Edited/extended table of VoIP status codes Status Meaning code The requested service is not available from the VoIP provider. Message is too long. URI is too long. Query format is not supported. URI is faulty. Incorrect ending Incorrect ending The requested service is not supported by the VoIP provider. The dialled number is temporarily unavailable. The recipient is not available. Double service query Too many "jumps": The query was rejected because the service server (proxy) has decided that this query has already passed through too many service servers. The maximum number is defined beforehand by the original sender of the query. Wrong number: In most cases this response means that you have simply omitted one or more digits in the phone number. The URI dialled is not unique and cannot be processed by the VoIP provider. The called party is busy. General faults: The call was cancelled before a call was established. The status code confirms receipt of the interruption signal. The server cannot process the query because the data entered in the media description is not compatible. The server notifies that the query will be processed as soon as a previous query has been completed. The server rejects the query because the phone cannot decrypt the message. The sender has used an encryption method that neither the server nor the receiver phone can decrypt. The proxy or the receiving device has discovered a fault while executing the query. It is therefore impossible to execute the query. If this occurs, the caller or the phone displays the fault and repeats the query after a few seconds. The number of seconds after which the query can be repeated may be transmitted to the caller or phone by the receiving device. The query cannot be processed by the recipient because the recipient does not have the functionality that the caller requires. If the recipient understands the query but does not process it because the sender does not have the necessary rights or the query is not permitted in the current context, status code 405 is transmitted instead of 501. In this case, the receiving device that transmits this error code is a proxy or a gateway and has received an invalid response from its gateway via which this query is to be processed.

493 500

Description of new and updated functions Edited/extended table of VoIP status codes Status Meaning code 503 The query cannot be processed by the receiving device or the proxy at present because the server is either overloaded or is being serviced. If it is possible for the query to be repeated in the foreseeable future, the server informs the caller or the phone of this. Time limit exceeded at the gateway. The server rejects the query because the indicated version number of the SIP protocol does not concur with at least the version that is used by the server or SIP device involved in this query. The server rejects the query because the message exceeds the maximum permitted size. The called party is busy. The called party has rejected the call. The called URI does not exist. The communication settings are not acceptable. The called party has hung up. VoIP socket error Connection cancelled because of timeout. Connection interrupted because of a SIP error. SIP memory error. SIP transaction memory error. Busy tone: No codec match between the calling and called party. General socket layer error. General socket layer error: Wrong socket number General socket layer error: Socket is not connected. General socket layer error: Memory error General socket layer error: Socket not available check IP settings/connection problem/VoIP setting incorrect. General socket layer error: Illegal application on the socket interface. No DNS server known. DNS name resolution failed. Insufficient resources for DNS name resolution. URL error.

504 505

924 925
Description of new and updated functions Deleted function: "Send line selection for outgoing calls with */#"
Deleted function: "Send line selection for outgoing calls with */#"
This section relates to the selection of default or non-default connections by adding # or * to the dialled number.
If your purchased telephone came with firmware version 02.140 or later already installed (Manufactured after May 2009), this function is not available. With these devices, it is no longer possible to select the non-default connection by adding an asterisk (*) to the dialled number or to select the default connection by adding a hash symbol (#). However, you can still use the line suffix to select the send line when dialling. If you add #0 to the number, it is dialled via the fixed line network. If you add #1, #2,., #6, the number is dialled via the corresponding VoIP connection. Further information about this can be found in the operating instructions for your telephone.
Dialling with the quick dial keys
If you have assigned a phone number to a number key on the handset as a quick dial number, it is dialled via the default connection if no line suffix is specified. Exception: A dialling plan has been defined for the number