Connections

You can establish up to 12 telephony connections (VoIP accounts) for your phone. 
Settings ->Telephony -> Connections

• On this screen you can
• See the status of the configured connections
• Activate or deactivate individual connections
• Define which of the connections are used by default for outgoing calls
• Change the configuration of connections or establish new connections
• Delete existing connections

Connections

The following information is shown for each configured connection:

Activating/deactivating connections

Only activated connections can be used for Internet telephony.

• To use a configured connection, select Active.

Specifying a default line for outgoing calls

• Select Default send account for the connection that you want to use as the default line for your phone calls. Only one can be selected.

Establishing new connections or editing existing ones

• Click on Edit in the row of a configured connection to change its configuration.
• Click on Edit in a row without a configured connection to establish a new connection. 

Configuring a VoIP account 

To configure VoIP accounts you need the relevant information about your provider for Internet telephony. Provider profiles with the general provider data are available to download from the Gigaset configuration server for the main VoIP providers. 

• In the Connection name or number field enter a name of your choice or the phone number for this connection.

Personal provider data
In both cases of manual configuration, with and without a provider profile, you now enter the per sonal registration data that you have received from your VoIP provider.

• Enter the following data:

• • Authentication name
  • Authentication password
  • User name
  • Display name

Depending on phone system further authentication data may be requested. 

Advanced settings

You can find further parameters for configuring your VoIP connection under Advanced settings. 

• Click on Show next to Advanced settings. 

Domain

Specify the last part of your SIP address (URI). 
Example: Example: For the SIP address 987654321@provider.de you would enter provider.de.

Proxy server address

The SIP proxy is your VoIP provider's gateway server. Enter the IP address or the DNS name of your SIP proxy server. Example: myprovider.com.

Proxy server port

Enter the number of the communication port that the SIP proxy uses to send and receive signal ling data (SIP port). Port 5060 is used by most VoIP providers.

Registration server

Enter the IP address or the DNS name of your registrar server. The registrar is needed when the phone is registered. It assigns your SIP address (username@domain) to the public IP address/port number your phone uses to log in. With most VoIP providers, the registrar server is identical to the SIP server. Example: reg.myprovider.de.

Registration server port

Enter the communication port used on the registrar. Port 5060 is used in most cases.

Registration refresh time (sec.)

Enter the time intervals at which the phone should repeat the registration with the VoIP server (SIP proxy) (a request will be sent to establish a session). The repeat is required so that the phone's entry in the tables of the SIP proxy is retained and the phone can therefore be reached. The repeat will be carried out for all activated VoIP phone numbers. The default is 180 seconds.
If you enter 0 seconds, the registration will not be repeated periodically. 

Network provider data 

The phone needs to know its public address in order to receive caller voice data.

The SIP protocol recognises the following options:

• The phone requests the public address from a STUN server on the Internet (Simple Transversal of UDP over NAT). STUN can only be used with asymmetric NATs and non-blocking firewalls.
• The phone does not direct the connection request to the SIP proxy but to an outbound proxy on the Internet that supplies the data packets along with the public address.

The STUN server and outbound proxy are used alternately to work around the NAT/firewall in the router/gateway. 

• Enter the required data for the STUN server or outbound proxy:

STUN enabled
Select Yes if you want your phone to use STUN as soon as it is used on a router with asymmetric NAT.

STUN server address
Enter the DNS name or the IP address of the STUN server on the Internet. If you have selected Yes in the STUN enabled field, then you must complete this field.

STUN server port
Enter the number of the communication port on the STUN server. The default port is 3478.

STUN refresh time (sec.)
Enter the time intervals (seconds) at which the phone should repeat the registration with the STUN server. The repeat is required so that the phone's entry in the tables of the STUN server is retained. The repeat will be carried out for all activated VoIP phone numbers. Ask your VoIP pro vider for the STUN refresh time if necessary. Default setting: 240 seconds.

NAT refresh time (sec.)
Specify the intervals at which you want the phone to update its entry in the NAT routing table. Specify an interval in seconds that is a little shorter than the NAT session timeout. As a rule you should not change the default value for the NAT update. Default setting: 20 seconds.

Outbound proxy mode
Specify when the outbound proxy should be used.

If you leave the Outbound server address field empty, the phone does not respond to the selected mode and operates as if Never were selected.

Outbound server address
Enter the DNS name or the IP address of your provider's outbound proxy. With many providers, the outbound proxy is identical to the SIP proxy.

Outbound proxy port
Enter the number of the communication port used by the outbound proxy. The default port is 5060.

DTMF in VoIP connections 

To send DTMF signals via VoIP, you must first define how key codes are to be converted into and sent as DTMF signals: as audible information via the speech channel or as a "SIP Info" message. 

Ask your provider which type of DTMF transmission is supported. 

Automatic negotiation of DTMF transmission
You have the following options: 

• If you activate the Yes option, the phone automatically attempts to set the appropriate DTMF sig nalling type for the current codec for each call. 
• If you activate the No option, you can use the other options to specify the DTMF signalling type. 

Send settings for DTMF transmission

Counting missed and accepted calls

Missed and accepted calls for this VoIP account are recorded in the call lists for the phone if this func tion is activated. 

• Select Yes for Missed/accepted calls count, if you wish to activate this function.

Allowing or blocking call waiting 

If you receive another incoming call during a call, this is indicated by Call Waiting by default. For each connection, it is possible to set whether or not Call Waiting is permitted.

• If you want to deactivate the function, select No. 

Setting ring tones

You can set ring tones for each configured VoIP account. You can specify different ring tones for exter nal and internal calls as well as for group calls, if this information is available for incoming calls (depending on the telephone system).

• Choose a Ring tone for all call types or different ring tones for specific call types.
• Click Test to play your chosen melody.

Saving settings

• Click on Save to save your settings for this connection.

Deleting a connection

• Click on Delete Connection to delete the displayed connection.

Audio settings

The voice quality of your VoIP calls is determined by the codec used for the transmission. To increase the quality, more data must be transmitted. Depending on the bandwidth of your DSL connection, this can then lead to problems with the volume of data – especially if two VoIP calls are made simul taneously – so that the transmission no longer takes place smoothly. The following settings allow you to adjust your Gigaset to your individual DSL connection.
Settings -> Telephony -> Audio
You can set the following parameters for the voice quality for each connection:

Time interval for RTP packets

• Select the interval for sending RTP packets (20 or 30 ms). 
  RTP (RTP = Real-Time Transport Protocol) is a protocol for the continuous transmission of audio visual data (streams) via IP-based networks. 

Voice quality

Both parties involved in a phone connection (caller/sender and recipient) must use the same voice codec. The voice codec is negotiated between the sender and the recipient when establishing a con nection. You can influence the voice quality by selecting (bearing in mind the bandwidth of your Internet connection) the voice codecs your phone is to use, and specifying the order in which the codecs are to be suggested when a VoIP connection is established.
ÛSelect the required codecs and define the sequence in which they should be used. 
The following voice codecs are supported by your phone:

Silence suppression

Silence suppression means that no data packets are sent during a pause in speaking. This means a lower data volume but call participants may interpret it as an interruption to the connection.

• Select No if you do not want silence suppression. Default setting: Yes

Saving settings

• Click on Save to save your settings on the screen.

Video settings

The video quality of your phone is determined by the resolution and the codec used for the transmission. 
Settings -> Telephony -> Video

• Select the resolution to be used for video transmission: internal for the integrated camera on the front panel, external for a camera connected to the device's USB port
• Select the required codecs and define the sequence in which they should be used. 

The following video codecs are supported by your phone:

Saving settings

• Click on Save to save your settings on the screen.

Call divert

For each configured VoIP account you can automatically forward incoming calls to another phone number.
Settings -> Telephony  -> Call Divert

• Specify for the VoIP account when a call should be diverted.

• Enter the       Phone number to which calls are to be forwarded.
• Click on Save to activate the call divert.

Local settings

Exceptions: Numbers for which you have set dialling plans.

• Select Other Country from the end of the Country list. 
• Enter the full prefix for the country in which you use your phone. Otherwise errors may occur with phone connections or during data exchange (e.g., between a fixed line network and a mobile net work).
  The prefix consists of the international prefix (International: Prefix and Code Number, e.g. 00 49 for Germany) and, if applicable, the prefix used for calls within the country (Local: Prefix and Code Number, e.g. 0 for national long-distance calls in Germany).

Setting country-specific ringback and dialling tones
Tones, e.g. dialling tone, ringback tone, busy tone or call waiting tone, vary from one country or region to another. You can choose from various tone groups for your phone. 
The Tone scheme is automatically determined on the basis of the country selected above. You can change the setting.

• Select the country or region whose tone scheme should be used for your phone.

Saving settings

• Click on Save to save your settings on the screen.

Dialling plans

You can use dialling plans to define which phone numbers should be called using which configured VoIP account and whether an area code should be dialled first.
Settings  -> Telephony -> Dialling Plans

• Enter the Phone number that the dialling plan is to apply to.
• Select Use area codes if you want to call this number with an area code.
• Select the connection that should be used to call this phone number from the drop-dow list.
• Enter a name for this dialling plan in the Comment field.
• Click on Add to add the rule to the list.
• Click on Delete to delete a rule from the list.
• Select Active to activate the rule.

Access Code
The access code you enter is automatically added as prefix to numbers during dialling.

• Define when it should be used. The prefix can be added if a call is initiated via
  • Outgoing calls list
  • Incoming calls list
  • LDAP
  • Local directory
  • Public net directory
  • Dial editor

Saving settings

• Click on Save to save your settings on the Dialling Plans screen.

Do Not Disturb (DND)

Block individual phone numbers and/or all anonymous calls. You will then be unavailable for these calls; your phone will not ring.
Settings -> Telephony -> Do Not Disturb

Call records

On this screen you define where call records should be saved.
You can start and replay call records within the CALL RECORDS area of the  Call list app. 

Settings -> Telephony -> Call Live Recording

Voice mail services

Some VoIP providers offer answer machines on the network – network mailboxes. These accept incoming calls on the corresponding VoIP phone number. 
You can access voice mails on the network mailbox within the VOICEMAIL area of the 
 Call list app.
To record all calls, set up a network mailbox for each of your VoIP accounts. 
Settings -> Telephony -> Voicemail Services