You can forward an incoming external call to the answer machine. Preconditions: There is sufficient free memory space. One of your handset's send/receive numbers matches a receive number of the answer machine. An external call is signalled on the handset: Press the display key. The answer machine starts immediately in recording mode and records the call. The set time for the call acceptance (page 79) is ignored. The display key will not be displayed if the memory is full.
Activating/deactivating two-way recording
You can record an external call with the answer machine. Precondition: You are conducting an external call. At least one of the handset's send/ receive numbers is assigned to the answer machine as a receive number (page 145).
Options Press the display key. Two Way Record Select and press OK. The two-way recording is indicated on the display by an advisory text and is added to the answer machine list as a new message. End Press the display key to stop the two-way recording. The maximum recording time depends on the memory available on the answer machine. If the memory is full, you will hear an end tone and the recording is aborted. The conversation recorded up to the end tone is then allocated to the answer machine list as a new message.
Inform the caller that the call is being recorded.
You can activate two-way recording for conference calls and toggling when making calls via a fixed line network.
Operating the base station answer machine Activating/deactivating call screening
Activating/deactivating call screening
During recording of a message you can screen a call via the handset loudspeaker. Precondition: The number called is assigned to the handset as a receive number (page 144). v Settings Voice Mail Local AM Call Screening Select and press OK ( = on) a Press and hold (idle status).
Permanently activating/deactivating call screening
Deactivating call screening for the current recording
You can deactivate the Call Screening function for the current call during recording via the handset. a Briefly press the end call key.
Setting up the answer machine
The answer machine has already been preset at the factory. Make individual settings using the handset.
Call acceptance/charge saving function for remote operation
You can set when you want the answer machine to accept a call. The options are: 0 sec., after 10 sec., 18 sec. or 30 sec. and the cost-saving setting Automatic. In Automatic mode, the following applies for call acceptance: u If there are no new messages, the answer machine accepts a call after 18 seconds. u If new messages are present, the answer machine accepts a call after 10 seconds. When operating remotely you can tell after approx. 15 seconds that there are no new messages (otherwise the answer machine would already have accepted your call). There will be no call charges if you hang up now. v Settings Voice Mail Local AM Ring Delay r Select 0 sec. / 10 sec. / 18 sec. / 30 sec. / Automatic. Save Press the display key. a Press and hold (idle status).

Ending a consultation call
Press the display key. You are reconnected with the external participant.
Initiating a conference call
Press the display key. You are in a three-way conference call with the external participant and the internal participant. If the internal participant who has been called ends the call (press end call key a), you will be connected with the external participant. If you press end call key a, the external call will be transferred to the internal participant.
Accepting/rejecting call waiting during an internal call
If you receive an external call during an internal call, you will hear the call waiting tone (short tone). With Calling Line Identification, the caller's number will appear in the display.
Ending an internal call, accepting an external call

Accept

Press the display key. The internal call is ended. You are connected to the external caller. Press the display key. The call waiting tone is turned off. You remain connected with the internal participant. The ringer tone can still be heard on other registered handsets.
Rejecting the external call

Reject

Using several handsets Using a handset as a room monitor
Using a handset as a room monitor
If the room monitor is activated, a previously stored destination number is called as soon as a set noise level is reached in the room. You can save an internal or external number in your handset as the destination number. The room monitor call to an external number stops after around 90 seconds. The room monitor call to an internal number (handset) stops after approx. 3 minutes (depending on the base station). During use as a room monitor, all keys are barred except for the end call key a. The speaker of the handset is muted. When the room monitor is activated, incoming calls to the handset are indicated without a ringer tone and are only shown on the screen. The display and keypad are not illuminated and advisory tones are also turned off. If you accept an incoming call, the room monitor is suspended for the duration of the call, but the function remains activated. If you deactivate then reactivate the handset, the room monitor remains activated.

Warning!

You should always check the operation of the room monitor before use. For example, test its sensitivity. Check the connection if you are forwarding the room monitor to an external number. When the function is switched on, the handset's operating time is severely reduced. If necessary, place the handset in the charging cradle. This ensures that the batteries do not run down. Ideally the handset should be positioned 1 to 2 metres away from the baby. The microphone must be directed towards the baby. The connection to which the room monitor is forwarded must not be blocked by an activated answer machine.

r Select volume. The setting will automatically be saved after approx. 3 seconds, if not then press the display key Save. If t is assigned with another function: Options Open menu. Volume Select and press OK. Configure setting (see above).
Setting the volume during a call: t Press the control key at the top.
Handset settings Changing ringer tones

Changing ringer tones

You can choose between five volumes (15; e.g. volume 2 = ) and the "crescendo" ring (6; volume increases with each ring = ). u Ringer tones: You can select various ringer tones. You can set different ringer tones for the following functions: u Ext. Calls: for external calls u Internal Calls: for internal calls u All: the same ringer tone for all functions

u Volume:

Settings for individual functions
Set the volume and melody depending on the type of signalling required. In the handset idle status: t Press the control key at the top. Ringer Settings Select and press OK. Ext. Calls / Internal Calls Select and press OK. Change multiple line input: r Set volume (16). s Scroll to the next line. r Select melody. Save Press the display key to save the setting. a Press and hold (idle status).
Same ringer tone for all functions
In the handset idle status: t Ringer Settings All Set volume and ringer tone (see "Settings for individual functions"). Save Press the display key to confirm the prompt. a Press and hold (idle status).
You can also adjust the ringer tones via Settings
Audio Settings Ringer Settings in the menu.
Handset settings Activating/deactivating advisory tones
Activating/deactivating the ringer tone
You can deactivate the ringer tone on your handset before you answer a call or when the handset is in idle status; the ringer tone can be deactivated permanently or just for the current call. The ringer tone cannot be reactivated while an external call is in progress.
Deactivating the ringer tone permanently

If you reply to the prompt with No, the display will not be repeated. The message New profile available will only be shown again if a newer version of the VoIP settings than the one rejected is available. You can deactivate the automatic version check via the Web configurator (page 161).
Entering/changing VoIP user data
The VoIP settings must also be extended for your personal data. You will receive all necessary data from your VoIP provider.
When making these entries, please remember the VoIP user data is case sensitive. To enter text see page 177.
v Settings Telephony VoIP (Enter system PIN) Provider Registr. Change multiple line input: Username: Enter the user name (caller ID) for your VoIP provider account. Username is often identical to your Internet telephone number (the first part of your SIP address, see page 137). Authent. Name: / Authent. Password: Enter the provider-dependent access data that has to be transferred by the phone to the SIP service at registration. Click Save to save the settings.
Making VoIP settings Setting the phone's IP address in LAN
Setting the phone's IP address in LAN
The base station requires an IP address in order to be "recognised" by the LAN. The IP address can be assigned to the base station automatically (by the router) or manually. u If performed dynamically, the router's DHCP server automatically assigns the base station an IP address. The base station's IP address can be changed according to router settings. u If performed manually/statically, you assign the base station a static IP address. This may be necessary depending on your network configuration.
For how to make the local network settings on the Web configurator, turn to page 131. To assign the IP address dynamically, the DHCP server on the router must be activated. Please also read the user guide for your router.
v Settings Base Local Network ~ Enter the system PIN and press OK. Change multiple line input: IP Address Type: Select Static or Dynamic. If you select Static, you must manually define the IP address and the subnet mask for the base station in the next lines, as well as the standard gateway and DNS server. IP Address: For IP Address Type = Dynamic: The IP address that is currently assigned to the base station will be displayed. It cannot be amended. For IP Address Type = Static: Enter the IP address that is to be assigned to the base station (overwriting the current settings). 192.168.2.2 has been preset. For the IP address see also page 194. Subnet Mask: For IP Address Type = Dynamic: The subnet mask that is currently assigned to the base station will be displayed. It cannot be amended. For IP Address Type = Static: Enter the subnet mask that is to be assigned to the base station (overwriting the current settings). The default setting is 255.255.255.0 For the subnet mask see also page 199.

Depending on the function selected, information or dialogue boxes are displayed in the working area, which allow you to make or change your phone settings.

Making changes

Make settings for entry fields, lists or options. u There may be restrictions regarding the possible values for a field, e.g. entering special characters or certain value ranges. u To open a list, click. You can choose between default values. u There are two kinds of options: Options in a list, from which you can activate one or several options. Active, i.e. selected options are highlighted with , non-active options with. You can activate an option by clicking. The status of the other options in the list does not change. You can deactivate an option by clicking.
Web configurator configuring the telephone via a PC Opening Web pages
Alternative options. The active option in the list is highlighted with , and the nonactive option with. You can activate an option by clicking. The previously activated option is deactivated. You can only deactivate an option by activating another option.

Applying changes

As soon as you have made your change on a page, activate the new setting on the phone by clicking Set. If your input in a field does not comply with the rules for this field, an appropriate error message will be displayed. You can then repeat the input.
Changes that have not been saved on your phone are lost if you move to another Web page or if the Web configurator is terminated, e.g. due to the time limit (page 126).
Buttons are displayed in the bottom section of the working area. Set Save entries on the phone. Cancel Reject changes made on the Web page and reload the settings that are currently saved in your phone to the Web page.

Opening Web pages

A brief outline of the navigation to the individual Web configurator functions is given below.

Example:

Setting DTMF signalling Settings Telephony Advanced Settings To open the Web page, carry out the following steps after registration: Click the Settings menu in the menu bar. Click the Telephony function in the navigation area. The subfunctions of Telephony are displayed in the navigation tree. Click the Advanced Settings subfunction. The Web page from Figure 2 will be shown in the Web browser.
Web configurator configuring the telephone via a PC Setting phone with Web configurator

The areas u General Provider Data (page 137) and u Network (page 138) can be shown and hidden by clicking the Show Advanced Settings and Hide Advanced Settings buttons. You must enter the VoIP provider's general access data in these areas. You can download this data for many VoIP providers from the Internet (see "Area: Auto Configuration").
Make the settings on the Web page. Save them in the phone, see page 139. Activate the connection if necessary, see page 140.

Area: IP Connection

Connection Name or Number Enter a name for the VoIP connection or the VoIP phone number (max. 16 characters). This name is used to display the connection on the handset and the Web configurator interface, e.g. during allocation of send and receive numbers (page 144), for the call display (page 35).

Area: Auto Configuration

The entire configuration process or a large part of the configuration for a VoIP connection is automated for many VoIP providers. You can download the necessary VoIP access data to your telephone from the Internet. You have the following options: u Fully automated configuration Preconditions: You have received an auto configuration code from your VoIP provider. The general access data for your VoIP provider is available for downloading. You can download all the data required for VoIP access from the Internet: Enter the auto configuration code you received from your VoIP provider in the Auto Configuration area in the Auto Configuration Code field. Click Start Auto Configuration. The telephone establishes a connection to the Internet and downloads all data required for the VoIP connection, i.e. the general provider information and your personal provider data (account data) are saved to your base station. If further information is used on the Web page, this is deleted as soon as Start Auto Configuration is clicked. The fields in the Personal Provider Data and General Provider Data areas and the server addresses in the Network area are overwritten by the downloaded data. Generally, you should not have to enter any additional data on this Web page.
If the message Download of settings not possible! File is corrupt! appears, no data will be loaded onto the phone. Possible causes of this are: The incorrect code has been entered (e.g. upper/lower case rules have not been followed). If necessary, enter the code again. The file that has been downloaded is invalid. Please consult your VoIP provider.
When the download is complete, the Connections list will be displayed. Activate the connection as described under page 140. You can now be reached on the corresponding VoIP phone number. u Automatic configuration of general VoIP provider data Precondition: You have received your account details from your VoIP provider (e.g. Authentication Name, Authentication password). Profile files of the most important VoIP providers are available on the Siemens Internet server for downloading. The address for the server is stored in your phone (page 159). Proceed as follows to load the data onto your telephone: In the Auto Configuration area, click Select VoIP Provider. This will display information on the download procedure.

Open the following Web page: Settings Telephony Number Assignment.
Web configurator configuring the telephone via a PC Assigning answer machine receive numbers (Gigaset C475 IP)
The display shows all registered handsets. A list is displayed for each handset showing the phone numbers that are configured and activated for the phone. The connection names are shown in the Connections column. The fixed line network connection is always at the end of the list. Define a VoIP phone number as the send number for each handset. To do this, click the option following the phone number in the for outgoing calls column. The previous assignment will automatically be deactivated.
The fixed line network number is permanently assigned to each handset as a send number. This assignment cannot be deactivated. It ensures that emergency calls can be made from every handset. The Gigaset.net number is also permanently assigned to each handset.
Select the phone numbers for each handset (fixed line network, VoIP) that are to be
assigned to the handset as receive numbers. To do this, click the option following the phone number in the for incoming calls column. Every handset can be assigned several phone numbers or no phone number ( = assigned). Now click Set to save your settings.
If a VoIP phone number that has been assigned to a handset as a send number is deleted, then the handset will automatically be assigned the first configured VoIP phone number. Calls made to a number that is not assigned to a handset as a receive number will not be signalled on any handset. If you have not assigned receive numbers to any of the handsets, calls to all connections will be signalled on all handsets.
Assigning answer machine receive numbers (Gigaset C475 IP)
You can specify for which of your phone numbers your telephone's answer machine is to accept the calls. To do this, assign receive numbers to the answer machine.
Once the new entry has been made, each VoIP phone number is assigned to the integrated answer machine as a receive number. If no receive number is assigned to the answer machine it will not accept any calls even if it is activated. For how to set and operate the answer machine, see page 73.
Open the following Web page: Settings Telephony Number Assignment. In the Answering machine area, select the phone numbers (fixed line network, VoIP) for

which the answer machine is to accept calls (if it is activated, see page 73). To do this, click the option following the phone number in the for incoming calls column. You can assign as many numbers as you wish to the answer machine ( = assigned). Now click Set to save your settings.
Web configurator configuring the telephone via a PC Call Forwarding to activate VoIP connections
Call Forwarding to activate VoIP connections
You can forward calls to your VoIP numbers and to your Gigaset.net number. You can forward calls to your VoIP numbers to any external number (VoIP, fixed line network or mobile network number). The forwarding is done via a VoIP connection. You can forward calls to your Gigaset.net number within the Gigaset.net, i.e. to another Gigaset.net number. You can define if and when calls to your Gigaset.net number and some of your VoIP numbers (VoIP account) should be forwarded to this VoIP number. You can also use the handset to define call forwarding and activate/deactivate it, see page 46. Open the following Web page: Settings Telephony Call Forwarding. The display shows a list of all your VoIP connections and your Gigaset.net number. Connections Select the name you have assigned to the VoIP number, or select Gigaset.net. When Select when a call to this VoIP number should be forwarded: When busy / No reply / Always. Select Off to deactivate call forwarding. Call number Enter the phone number to which the calls should be forwarded. Please note that you may have to enter the area code when forwarding to a fixed line network number in the same area (depending on your VoIP provider and the setting for the automatic area code, see page 148). The settings only affect the selected phone number.
For how to forward your fixed line network number, see page 45.
Setting DTMF signalling for VoIP
DTMF signalling is required, for example, for querying and controlling certain network mailboxes via digit codes or for remote operation of the integrated answer machine (Gigaset C475 IP). To send DTMF signals via VoIP you must first define how key codes should be converted into and sent as DTMF signals: as audible information via the speech channel or as "SIP Info" message. Ask your VoIP provider which type of DTMF transmission it supports. Open the following Web page: Settings Telephony Advanced Settings. In the DTMF over VoIP connections area, make the required settings for sending DTMF signals. Activate Audio or RFC 2833, if DTMF signals are to be transmitted acoustically (in voice packets). Activate SIP Info if DTMF signals are to be transmitted as code. Now click Set to save your settings. 146

u You want to block your phone for all 0900 numbers.
Dialling plan: Phone Number = 0900 Connection Type = Block u All calls to the mobile phone network shall be made via your VoIP connection with provider B. Dialling plans: Phone Number = 017 Connection Type = IP3, provider B and the corresponding entries for "015" and "016". Version 2.1, 08.01.2007
Activating/deactivating dialling plans
plan ( = activated). A deactivated dialling plan will not take effect until it is reactivated. 150
Click the option in the Active column to activate/deactivate the corresponding dialling
Web configurator configuring the telephone via a PC Activating/deactivating network mailbox, entering numbers

Deleting dialling plans

The dialling plan is deleted from the list immediately. The space in the list is released.
Predefined dialling plans set as defaults (for emergency numbers cannot be deactivated and cannot be deleted.
Click Delete next to the dialling plan you wish to delete.
Dialling plans for emergency numbers (e.g. the local emergency service number) are factory-set for certain countries. The fixed line network is determined as the Connection Type. These dialling plans cannot be deleted, deactivated or blocked. However, you can change the Connection Type. This should only be changed if the phone is not connected to the fixed line network. If you choose a VoIP connection, please make sure the VoIP provider supports emergency calls. If no emergency numbers are set by default, you should define dialling plans for emergency numbers yourself and assign them to a connection of which you know that it supports emergency calls. Emergency calls are always supported by fixed line networks.
Emergency numbers cannot be dialled if the keypad lock is activated. Before dialling, press and hold the hash key #, to release the keypad lock. If you have activated an automatic area code (page 148) and if no dialling plan for emergency calls is defined, the area code will also be prefixed to emergency calls made via VoIP.
Many fixed network providers and VoIP providers offer answer machines on the network these are know as network mailboxes. Each network mailbox accepts incoming calls made via the corresponding line (fixed line network or corresponding VoIP phone number). You can enter the relevant network mailbox for each configured connection (VoIP, fixed line network) via the Web configurator. You can activate or deactivate the network mailbox for your VoIP connections.

The phone does not dial an entered number. The display shows Not possible!. The number may be blocked (dial rule). Open the Dialling Plans Web page of the Web configurator and delete or deactivate the block. When establishing a connection, check the phone's local IP address that has been entered. You can check the IP address on your handset. Check the LAN connections for the PC and phone. Check that your phone can be reached. Send a ping command, e.g. from your PC, to the phone (ping s <local IP address of the phone>). You have tried to reach the phone via a secure http (https://.). Try again with http://.
You cannot establish a connection to the phone with your PC's Web browser.
Appendix Questions and answers You cannot be reached for calls from the Internet.
There is no entry for your phone in your router's routing table. Check the settings for the NAT refresh time (page 139). Your phone is not registered with the VoIP provider. You have entered the wrong user ID or an incorrect domain (page 137).
No firmware update or VoIP profile download is carried out. 1. If Currently not possible! is displayed, the VoIP connections may be busy or a download/update is already being carried out. Repeat the process at a later time.
2. If File corrupted! is displayed, the firmware or profile file may be invalid. Please only use firmware and downloads that are made available on the preconfigured Siemens server (page 159) or at www.siemens.com/gigasetcustomercare. 3. If Server not accessible! is displayed, the download server may not be accessible. The server is currently not accessible. Repeat the process at a later time. You have changed the preconfigured server address (page 159). Correct the address. If necessary, reset the base station. 4. If Transmission Error XXX is displayed, an error has occurred in the transmission of the file. An HTTP error code is displayed for XXX. Repeat the process. If the error occurs again, consult the Service department. 5. If Check IP settings! is displayed, your phone may not be connected to the Internet. Check the cable connections between the phone and router and between the router and the Internet. Check whether the phone is connected to the LAN, i.e. it can be reached at its IP address. You cannot listen to or control the network mailbox. VoIP: Your VoIP provider does not support the type of DTMF signalling set up on your phone. Ask your VoIP provider which signalling it supports and change the settings on your phone (page 146) if necessary. Operating the base station within a PABX: Your PABX is set for dial pulsing. Set your PABX to touch tone dialling.

Preamble

Appendix Gigaset C470 IP/C475 IP Free software For example, on rare occasions, there may be a special need to encourage the widest possible use of a certain library, so that it becomes a de-facto standard. To achieve this, non-free programs must be allowed to use the library. A more frequent case is that a free library does the same job as widely used non-free libraries. In this case, there is little to gain by limiting the free library to free software only, so we use the Lesser General Public License. In other cases, permission to use a particular library in non-free programs enables a greater number of people to use a large body of free software. For example, permission to use the GNU C Library in non-free programs enables many more people to use the whole GNU operating system, as well as its variant, the GNU/Linux operating system. Although the Lesser General Public License is Less protective of the users' freedom, it does ensure that the user of a program that is linked with the Library has the freedom and the wherewithal to run that program using a modified version of the Library. The precise terms and conditions for copying, distribution and modification follow. Pay close attention to the difference between a "work based on the library" and a "work that uses the library". The former contains code derived from the library, whereas the latter must be combined with the library in order to run.
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION 0. This License Agreement applies to any software library or other program which contains a notice placed by the copyright holder or other authorized party saying it may be distributed under the terms of this Lesser General Public License (also called "this License"). Each licensee is addressed as "you". A "library" means a collection of software functions and/or data prepared so as to be conveniently linked with application programs (which use some of those functions and data) to form executables. The "Library", below, refers to any such software library or work which has been distributed under these terms. A "work based on the Library" means either the Library or any derivative work under copyright law: that is to say, a work containing the Library or a portion of it, either verbatim or with modifications and/or translated straightforwardly into another language. (Hereinafter, translation is included without limitation in the term "modification".) "Source code" for a work means the preferred form of the work for making modifications to it. For a library, complete source code means all the source code for all modules it contains, plus any associated interface definition files, plus the scripts used to control compilation and installation of the library. Activities other than copying, distribution and modification are not covered by this License; they are outside its scope. The act of running a program using the Library is not restricted, and output from such a program is covered only if its contents constitute a work based on the Library (independent of the use of the Library in a tool for writing it). Whether that is true depends on what the Library does and what the program that uses the Library does. 1. You may copy and distribute verbatim copies of the Library's complete source code as you receive it, in any medium, provided that you conspicuously and appropriately publish on each copy an appropriate copyright notice and disclaimer of warranty; keep intact all the notices that refer to this License and to the absence of any warranty; and distribute a copy of this License along with the Library. You may charge a fee for the physical act of transferring a copy, and you may at your option offer warranty protection in exchange for a fee. 2. You may modify your copy or copies of the Library or any portion of it, thus forming a work based on the Library, and copy and distribute such modifications or work under the terms of Section 1 above, provided that you also meet all of these conditions: a) The modified work must itself be a software library. b) You must cause the files modified to carry prominent notices stating that you changed the files and the date of any change. c) You must cause the whole of the work to be licensed at no charge to all third parties under the terms of this License. d) If a facility in the modified Library refers to a function or a table of data to be supplied by an application program that uses the facility, other than as an argument passed when the facility is invoked, then you must make a good faith effort to ensure that, in the event an application does not supply

Upgrade your Gigaset to a cordless PABX:

Gigaset C47H handset

u u u u u u u u u
Illuminated graphical colour display (65k colours) Illuminated keypad Handsfree talking Polyphonic ringer tones Directory for around 150 entries SMS (precondition: CLIP must be enabled) Headset socket Room monitor www.siemens.com/gigasetC47H

Gigaset S67H handset

u u u u u u u u u u
HDSP ready Illuminated graphical colour display (65k colours) Illuminated keypad Handsfree talking Polyphonic ringer tones Directory for around 250 entries Picture CLIP SMS (precondition: CLIP must be enabled) Headset socket Room monitor www.siemens.com/gigasetS67H

Gigaset SL37H handset

HDSP ready Illuminated graphical colour display (65k colours) Illuminated keypad Handsfree talking Polyphonic ringer tones Directory for around 250 entries Picture CLIP SMS (precondition: CLIP must be enabled) PC interface, e.g. for managing directory entries, ringer tones and screensavers u Headset socket u Bluetooth u Room monitor u Walky-talky function www.siemens.com/gigasetSL37H

Gigaset S45 handset

u u u u u u u u
Illuminated colour display (4096 colours) Illuminated keypad Handsfree talking Polyphonic ringer tones Directory for around 150 entries SMS (precondition: CLIP must be enabled) Headset socket Room monitor www.siemens.com/gigaset

Gigaset repeater

The Gigaset repeater can be used to increase the reception range of your Gigaset handset to the base station. www.siemens.com/gigasetrepeater
Gigaset HC450 door intercom for cordless phones
u Use the intercom from your cordless phone
no need for a fixed home phone
u Intuitive user functions using the display keys u u u u u u
(open door, switch on entry light) Simple to configure using the handset menu Forward to outside phone numbers (intercom feature) Simple to install and register with the Gigaset system Replaces existing call button no further cable is needed Supports the existing doorbell and standard door opener Configuration options for the second ringer key (separate intercom call, activating interior lighting, or function such as first ringer key) www.siemens.com/GigasetHC450
All accessories are available from your mobile phone retailer. Only use original accessories. This will avoid possible health risks and damage to property, and also ensure that all the relevant regulations are complied with.
Gigaset C470-475 IP / EN for IM-Ost / A31008-xxxx-xxxx-x-xxxx / glossary.fm / 18.12.07

Glossary

Asymmetric Digital Subscriber Line

Global System for Mobile Communication
Originally, European standard for mobile networks. GSM can now be described as a worldwide standard. In the USA and Japan national standards are now more frequently supported than in the past. 193
Headset Combination of microphone and headphone. A headset makes handsfree talking more comfortable. There are headsets available that are connected to the handset by a cable. HTTP Proxy Server via which the Network subscribers can process their Internet traffic. Hub Uses one Infrastructure network to connect several Network subscribers. All data sent to the hub by one network subscriber is forwarded to all network subscribers. See also: Gateway, Router.
Institute of Electrical and Electronics Engineers
International body that defines standards in electronics and electro-technology, concerned in particular with the standardisation of LAN technology, transmission protocols, data transfer rate and wiring. Infrastructure network Network with central structure: all Network subscribers communicate via a central Router. Instant messaging Service that uses a client program to allow chatting in real time, i.e. to send brief messages to other subscribers on the Internet. Internet Global WAN. A series of protocols have been defined for exchanging data, known by the name TCP/IP. All Network subscribers are identifiable via their IP address. DNS assigns a Domain name to the IP address. Important services on the Internet include the World Wide Web (WWW), e-mail, file transfer and discussion forums. Internet Service Provider Enables access to the Internet for a fee. IP (Internet Protocol) TCP/IP protocol on the Internet. IP is responsible for addressing subscribers in a Network using IP addresses and routes data from the sender to the recipient. IP determines the paths (routing) along which the data packets travel. IP address A unique address for a network component within a network based on the TCP/IP protocols (e.g. LAN, Internet). On the Internet, domain names are usually assigned instead of IP addresses. DNS assigns the corresponding IP address to the domain name. The IP address has four parts (decimal numbers between 0 and 255) separated by full stops (e.g. 230.94.233.2).
The IP address is made up of the network number and the number of the Network subscribers (e.g. phone). Depending on the Subnet mask, the front one, two or three parts make up the network number and the rest of the IP address addresses the network component. The network number of all the components in any one network must be identical. IP addresses can be assigned automatically with DHCP (dynamic IP addresses) or manually (static IP addresses). See also: DHCP. IP pool range Range of IP addresses that the DHCP server can use to assign dynamic IP addresses.

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.

Dial Number

12345678
If you press Yes, the number is dialled. If you press No, the page with the hyperlink is displayed again. The number will not

be dialled. Or:

Entering text
If necessary, use q to navigate to the line containing the field into which you want to Enter your text using the handset keys (for information on entering text see the appendix to the user guide belonging to your phone). If necessary, navigate to other text fields to complete them or make a selection Press the right display key to complete the entry and send the data.

Making selections

(see below). enter text. The cursor flashes in the text field.
If necessary, use q to navigate to the line, in which you would like to make a selection. Press left or right on the control key several times to make the desired selection. Use q to navigate to other selection fields and make your selection as described above. Press the left display key to complete the selection and send the data.

Setting options

Use q to navigate to the line containing the option. The line is highlighted. Activate or deactivate the option via the control key v (press right) or the right display If necessary, navigate to other options or text fields to set or complete them. Press the left display key (e.g. Send) to complete the entry and send the data.
Version 4, 29.10.2007 key (e.g. OK).
Description of new and updated functions Sending and receiving SMS (text messages) via VoIP
Sending and receiving SMS (text messages) via VoIP
This section amends the chapter "SMS (text messages)" in the user guide for your Gigaset VoIP phone.
You can now use your telephone to send and receive SMS messages via the fixed line network and VoIP. You can receive SMS messages (abbreviated: SMS) via all of your telephone's connections (with the exception of Gigaset.net). You must explicitly specify the (send) line via which the SMS messages are to be sent (fixed line network or one of your VoIP connections). Precondition: Your fixed line network and VoIP providers support SMS functionality.

Setting the send line

You define the send line when setting the SMS centres. For each individual SMS centre, you can specify which of your lines is to be used to send SMS messages when this SMS centre is activated as the send centre. Please note Before you specify one of your VoIP connections as the send line, check with your VoIP provider whether the SMS centre can be reached via the VoIP connection. Not all VoIP providers support special phone numbers! v Messaging SMS Settings Service Centres s Select SMS centre (e.g. Service Centr. 1) and press OK. s Scroll to the Send via line to select the send line to be used when sending SMS messages via this SMS centre. The fixed line network is preset by default. Edit Press the display key. A list of your phone's connections will be displayed. You can select from your fixed line network connection and all VoIP connections that you have configured. The standard names for the connections are displayed. IP1 to IP6, Fxd. ln. Fxd. ln. / IP1 / IP2 /. Select the VoIP or fixed line connection and press OK. Save Press the display key to save the changes.

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

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.
Rejecting a waiting call during a VoIP call
You can now reject a waiting call while conducting a conversation via VoIP. You are conducting an external call via a VoIP connection. A second caller (waiting call) is signalled: Options Reject waiting call Select the above and press OK to reject the waiting call.
Description of new and updated functions Operating the base on on the PABX Setting access codes (external line prefixes)
Operating the base on on the PABX Setting access codes (external line prefixes)
This section amends the chapter "Operating the base on the PABX" in the user guide for your Gigaset VoIP phone.
Depending on your PABX, you must dial an access code before making external calls in order to obtain an external line. You can store this access code in your phone. For example, the access code is then automatically placed before the numbers selected from the calls list. v Settings Base Add. Features Access Code Select and press OK. ~ Enter or edit the access code (maximum three digits) and press OK. a Press and hold (idle status). If an access code is set, the following applies: u The access code is added automatically when dialling from the calls list/answering machine list and when dialling emergency numbers and SMS centre numbers. u When dialling numbers manually and dialling numbers from the directory you must add the access code yourself.

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.

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"
This section amends the section "Using the directory and lists Using online directories" in the user guide for your Gigaset VoIP phone.
Entering the town/city name when searching for an entry
To avoid repeated entries, the names of the last five towns/cities entered are displayed in the City field. You can select the displayed name of the town/city with s and confirm with OK or You can enter a new name.
Description of new and updated functions Amendment to "Changing the display language"
Amendment to "Changing the display language"
The section amends the section "Handset settings Changing the display language" in the user guide for your Gigaset VoIP phone.
Parts of the menu are not displayed in the language selected.
.and three or more handsets are registered on your base. The language set on at least three handsets is not a standard language of the base. The standard base languages are: English, French, German, Italian, Spanish, Portuguese and Dutch. On your base, display texts are only stored for the standard languages. In addition, these display texts can be stored in the base in two other languages or in another language for two different types of Gigaset handsets. When selecting the language on the handset, these texts are downloaded to the base from the Internet. If another non-standard language is set on a third handset, some display texts appear in one of the standard languages on this handset. Both of the non-standard languages are saved on the base, which are set with the lower number of internal numbers. If there is no further handset registered on the base whose type and language setting correspond to an additionally loaded language, then the memory is freed up. If necessary, the language set for another registered handset is loaded onto the base.

In general no special telephone or router configuration is required when operating a Gigaset VoIP phone with a NAT router. The configuration settings described in this section are only necessary if you encounter one of the following problems.
Typical problems caused by NAT
u No incoming calls are possible via VoIP. Calls to your VoIP phone number are not put

through.

u Outgoing calls via VoIP are not connected. u A connection is established with the other party, but you cannot hear them and/or they

cannot hear you.

Possible solution
1. Change the port numbers of the communication ports (SIP and RTP ports) on your telephone ( "1. Changing the port numbers for SIP and RTP on your VoIP phone"). 2. In some cases, you must also define port forwarding for the telephone's communication ports on the router ( "2. Setting port forwarding on the router").
1. Changing the port numbers for SIP and RTP on your VoIP phone
On your VoIP telephone, define different (local) port numbers for the SIP and RTP ports (between 1024 and 49152). u These numbers must not be used by any other application or host in the LAN and u be considerably higher or lower than the SIP and RTP port numbers that you usually use (and are preset on the phone). This procedure is particularly useful if additional VoIP phones are connected to the router.
To change the SIP and RTP port numbers on your VoIP phone, proceed as follows:
Connect your PC's browser to the Web configurator of the telephone and log in ( user guide for your phone). Open the Web page Settings Telephony Advanced Settings and change the settings for the SIP and RTP ports ( Page 23).
To help you remember the new port numbers (e.g. for router configuration), you can choose numbers that are very similar to the standard settings, e.g. SIP port RTP port Version 4, 29.10.49004 to 49010 instead of instead of to 5010
Save the changes on your telephone. Wait for the active VoIP connections to be re-registered. To do so, switch to the Web page Settings Telephony Connections to see the Status of your VoIP connections. Check to see whether the problem persists. If it does, perform step 2.
2. Setting port forwarding on the router
To ensure that your specified SIP and RTP port numbers are used on the WAN interface with the public IP address, you must define port forwarding rules for the SIP and RTP ports on the router.
To define port forwarding on the router, proceed as follows:
The terms used in the following can vary from router to router. To forward a port, you must make the following specifications (example):

Protocol

UDP UDP

Public port

49060 4900449010

Local port

Local host (IP)
192.168.2.10 192.168.2.10 for SIP for RTP
Protocol Enter UPD as the protocol to be used. Public port Port number/port number range on the WAN interface Local port The SIP and RTP port numbers set on the telephone. In the new firmware version for Gigaset VoIP telephones, you can set a RTP port range. In this case, you must also define corresponding port forwarding for this range. Local host (IP) Local IP address of your phone in the LAN. You can see the phone's current IP address in the handset display by pressing the paging key on the base. To enable the router to perform this port forwarding, the DHCP settings of the router must ensure that the telephone is always assigned the same local IP address i.e. the DHCP does not change the IP address assigned to the telephone during operation. Alternatively, you can assign a fixed (static) IP address to the telephone ( user guide for your phone). However, you must ensure that this IP address is not within the address range reserved for DHCP and is not assigned to any other LAN subscriber.
Description of new and updated functions Edited/extended table of VoIP status codes
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
Gigaset IP Phones / IM-NORD EN / A31008-xxxx-xxxx-x-xxxx / HS_menutrees.fm / 3/20/09
Handset menu overviews Gigaset S67H to Gigaset S675 IP, Gigaset S68H to Gigaset S685 IP

Handset menu overviews

Gigaset S67H to Gigaset S675 IP, Gigaset S68H to Gigaset S685 IP
New and updated menus and submenus are marked in orange. Please note that a few digit combinations (shortcuts) for quick entry to the submenus have also changed. They are also marked in orange. 1 Messaging An SMS mailbox (general or private) activated without a PIN 1-1-1 New SMS 1-1-2 Incoming (0) 1-1-3 Draft (0) An SMS mailbox activated with a PIN or 2-3 mailboxes 1-1-1 Mailbox 1-1-1-1 New SMS 1-1-1-2 Incoming (0) 1-1-1-3 Draft (0) 1-1-2 Mailbox 1 to Mailbox 2 1-1-4 Mailbox 3 1-1-2-1 New SMS to 1-1-4-1 1-1-2-2 Incoming (0) to 1-1-4-2 1-1-2-3 Draft (0) to 1-1-4-3 1-1-5 SMS Service 1-1-6 Settings 1-1-6-1 Service Centres 1-1-6-2 SMS Mailboxes 1-1-6-3 Notify Number 1-1-6-4 Notify Type 1-1-6-5 Status Report Version 4, 29.10.2007 1-2 E-mail 1-1 SMS

1-3 Messenger

1-3-1 Buddies 1-3-2 User Status 1-3-2-1 Change Status 1-3-2-2 Info 1-3-3 Messages

Sel. Services

2-1 Info Center 2-2 VoIP

Page 8

2-2-6 Call Diversion 2-2-7 Call Waiting

2-3 Fixed Line

2-3-6 Call Diversion 2-3-7 Call Waiting
*) Menu item Withhold No. is no longer available. It is replaced by 2-6 Next Call.
2-4 Ringback Off 2-5 Always anon. 2-6 Next Call Calls List Add. Features 4-4-2 Bluetooth 4-4-3 Directory 4-6 Missed Appts. 7 Alarm Clock Calendar Resource Dir.

Only for Gigaset S68H

4-3 Room Monitor 4-4 Data Transfer
7-1 Screensavers 7-2 Caller Pictures 7-3 Sounds 7-4 Capacity Version 4, 29.10.2007

Settings

8-1 Date/Time 8-2 Audio Settings 8-2-1 Handset Volume 8-2-2 Ringer Settings 8-2-2-1 Ext. Calls 8-2-2-2 Internal Calls 8-2-2-3 Appointments 8-2-2-4 All 8-2-3 Advisory Tones 8-3 Display 8-3-1 Screen Saver 8-3-2 Colour Scheme 8-3-3 Contrast 8-3-4 Backlight 8-4 Handset 8-4-1 Language 8-4-2 Auto Answer 8-4-3 Register H/Set 8-4-4 Select Base 8-4-5 Area Codes 8-4-6 Reset Handset 8-5 Base 8-5-1 Calls List Type 8-5-1-1 Missed Calls 8-5-1-2 All Calls 8-5-2 Music on hold 8-5-3 System PIN 8-5-4 Base Reset 8-5-5 Add. Features 8-5-5-1 Repeater Mode 8-5-5-2 Access Code 8-5-5-3 Eco Mode 8-5-6 Local Network 8-5-8 Software Update

Page 18

8-6 Voice Mail

8-6-1 Local AM

8-6-1-1 Ans Machine 8-6-1-2 Call Screening 8-6-1-3 Announcements 8-6-1-4 Message Length 8-6-1-5 Recording Quality 8-6-1-6 Ring Delay

8-6-2 Network AM(s)

8-6-2-1 Net AM Fxd. ln. 8-6-2-2 Net AM IP1 :
(dependent on the number of configured VoIP phone numbers and receive numbers on the handset)
8-6-2-7 Net AM IP6 8-6-3 Set Key 1 Local AM Net AM Fxd. ln. Net AM IP1
Net AM IP6 8-7 Telephony 8-7-1 Default Line 8-7-1-1 VoIP 8-7-1-2 Fixed Line 8-7-2 Connection Assist. 8-7-6 Fixed Line 8-7-7 VoIP 8-7-6-2 Recall

Enter system PIN

Show Stat. on HS Select Provider Provider Registr.
Handset menu overviews Gigaset C47H to Gigaset C470/C475 IP
Gigaset C47H to Gigaset C470/C475 IP
New and updated menus and submenus are marked in orange. Please note that a few digit combinations (shortcuts) for quick entry to the submenus have also changed. They are also marked in orange. 1 Messaging An SMS mailbox (general or private) activated without a PIN 1-1-1 New SMS 1-1-2 Incoming (0) 1-1-3 Draft (0) An SMS mailbox activated with a PIN or 2-3 mailboxes 1-1-1 Mailbox 1-1-1-1 New SMS 1-1-1-2 Incoming (0) 1-1-1-3 Draft (0) 1-1-2 Mailbox 1 to Mailbox 2 1-1-4 Mailbox 3 1-1-2-1 New SMS to 1-1-4-1 1-1-2-2 Incoming (0) to 1-1-4-2 1-1-2-3 Draft (0) to 1-1-4-3 1-1-5 SMS Service 1-1-6 Settings 1-1-6-1 Service Centres 1-1-6-2 SMS Mailboxes 1-1-6-3 Notify Number 1-1-6-4 Notify Type 1-1-6-5 Status Report 1-2 E-mail 1-3 Messenger Version 4, 29.10.2007 1-3-1 Buddies 1-3-2 User Status 1-3-2-1 Change Status 1-3-2-2 Info 1-3-3 Messages 39