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Using the telephone (page
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Red predeterminada Default network is used.
VPN VPN is used.
RegistradorImportant: Every port opening on the router is a security risk. Perform the appropriate security measures.
That is why you may need to set up port forwarding for incoming SIP packets in the router for the Puerto SIP set up in the provider's/PBX's configuration.
Puerto de registrador (1...65535) 
Proxy de salidaDesactivado: Disables the outbound proxy.
Asignar automáticamente: Automatically determines an outbound proxy.
Asignar manualmente: Enables you to enter an outbound proxy.
If the Asignar manualmente option has been selected, additonal entries must be made:
Puerto del proxy de salida (1...65535)
Note: You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
Puerto SIP (1...65535)
You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
Note: The enabling of the SIP session timer can cause increased breaks in conversations after the set interval, if a provider has not implemented the renewal of the session cleanly. In this case, you should disable the SIP session timer.
Protocolo de transporte SIPImportant: If encryption is enabled by SIPS, the TCP transport protocol is used. Manual settings are overwritten.
UDP: UDP is used. TCP: TCP is used.|
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Servidor SIP por defecto: Server for the normal system telephone operation.
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Servidor Broadsoft: Server for providers with Broadsoft certification.
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ESTOS CSTA (ECSTA): Server for the use of uaCSTA.
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Important: If you are using this on the PBX, the PBX specifies the encryption method. The PBX overwrites manual settings.
Note: If you have saved a root certificate for encrypting calls in your PBX, this is uploaded automatically to the telephone. You must enter the root certificate fingerprint in the telephone so that the root certificate can be verified.
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SIPS: Encryption of SIPS via TLS.
SIP/TLS: Encryption of SIP via TLS.
SIP NAT-Traversal|
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STUN server set up for the provider/PBX, if the Activado con uso de STUN option has been selected
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Important: Every port opening on the NAT router is a security risk. Perform the appropriate security measures.
Activado : NAT traversal is performed by the provider/PBX. The SIP request also contains an "rport" part in which the IP addresses used here (local, public) are transported.
: NAT traversal is performed by the provider/PBX. The SIP request also contains an "rport" part in which the IP addresses used here (local, public) are transported.Desactivado : NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.
: NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.Activado con uso de STUN : NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a Servidor STUN under Ajustes.
: NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a Servidor STUN under Ajustes.Note: If Activado is selected: if there are problems with unilateral call connections, a STUN server should be used (enabled when you are using STUN).
RTP NAT-Traversal|
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STUN server set up for the provider/PBX, if the Activado con uso de STUN option has been selected
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Important: Every port opening on the NAT router is a security risk. Perform the appropriate security measures.
Desactivado : NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.
: NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.Activado con uso de STUN : NAT traversal is performed by the provider/PBX. To do this, you must also specify a STUN server.
: NAT traversal is performed by the provider/PBX. To do this, you must also specify a STUN server. IP/URL del servidor STUN Puerto del servidor STUN (1...65535) STUN server query interval (min) (1...60)Note: If there are problems with unilateral call connections, enabling the RTP might help.
Important: Each port forwarding is a security risk. For this reason, we recommend you use this function only if absolutely necessary.
NAT-Keep-AliveNote: Some providers lock accounts if the Intervalo NAT-Keep-Alive (s) is too short. As a rule, this is reported with error message 503 during SIP registration. If this problem occurs, we recommend you reduce the value (e.g. to 180).
Máxima calidad : Codec G.722 has priority level 1, Codec G.711 has priority level 2 and Codec iLBC has priority level 3.
: Codec G.722 has priority level 1, Codec G.711 has priority level 2 and Codec iLBC has priority level 3.Máxima compresión : Codec iLBC has priority level 1, Codec G.722 has priority level 2 and Codec G.711 has priority level 3.
: Codec iLBC has priority level 1, Codec G.722 has priority level 2 and Codec G.711 has priority level 3.Forzar G711 : Codec G.711 is required.
: Codec G.711 is required.|
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int Auerswald PBX sub system IPv4 * profile not selected
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Note: If your are unsure which DMFT signalling type you must select, you cand find out the right type by trying them out.
Fuera de banda con tono de confirmación local según RFC2833 : The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data. You hear a confirmation tone to confirm that the DTMF signals have been sent successfully.
: The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data. You hear a confirmation tone to confirm that the DTMF signals have been sent successfully.Inband, DTMF signals sent through the audio channel : The telephone uses the same channel (DTMF tones) to transfer DTMF signals and voice data.
: The telephone uses the same channel (DTMF tones) to transfer DTMF signals and voice data.Fuera de banda según RFC2833 : The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data.
: The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data.Búfer de jitter (ms) (40...160)
Lower values: A lower number of RTP packets can be cached in order to buffer disruptions or to compensate for them.
Greater values: A greater number of RTP packets can be cached in order to buffer disruptions or to compensate for them.
Important: If you are using this on the PBX, the PBX specifies the encryption method. The PBX overwrites manual settings.
Desactivado: Forces de-activation of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in the sub-system operation, external VoIP subscriber) requires call data to be encrypted by SRTP.
Preferido: Switches on negotiation for the encryption of call data by SRTP. When a call is made, the call partner will be asked if encryption is possible. If so, the call data will be encrypted and then transmitted, if not, the call data will not be encrypted.
Obligatorio: Forces activation of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in the sub-system operation, external VoIP subscriber) does not support the encryption of call data by SRTP.
Utilización de números de línea externa|
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Set type Centralita
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Selected type Centralita (page Settings > Providers & PBXs > Provider)
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to display the list of external contacts and dial an external account (public exchange). Tap on 
to display the list of internal contacts. Only internal telephone numbers can be dialled. Important: If operated as a system telephone on the PBX, the keypad sequences are predefined by the PBX. The PBX overwrites any keypad sequences you have entered manually.
Note: When selecting the type Proveedor VoIP (internet) under Ajustes > Proveedores y centralitas > Configurar > Proveedor > Tipo some keypad sequencens and the exchange line access number are not available.
Captura: The keypad sequence is required to carry out pick-ups on PBXs where a pick-up sequence is mandatory for this function. Information from SIP notify messages are not taken into consideration.
Aviso InterCom: The keypad sequence is required to make an InterCom announcement. InterCom announcements can also be made on accounts where the telephone is not used as system telephone.
Manos libres InterCom: The keypad sequence is required to enable Intercom hands-free calling at a destination which does not support automatic call acceptance.
Confirmación de alarmas: The keypad sequence is required to acknowledge an alarm call with the keypad.
Cola de espera: For moving into the waiting loop.
Predeterminado: Selects an evaluation method for phone numbers which will work for most providers.
Standard (use From headers as screened-number): Selects an evaluation method for phone numbers which will work for most providers. The user provided number (corresponds to CLIP no screening) of the header “From“ will be taken over into the network provided number. The phone numbers can, for example, be used for Call Through via VoIP.
Caution: If you have selected this option, your telephone and your PBX are vulnerable to attacks, for example by transmission of a phone number manipulated by the attacker. Therefore, to protect your telephone and PBX, assure yourself that the VoIP provider has integrated sufficient security measures in the number presentation.
Tal y como se describe en RFC3325: Selects a phone number evaluation method in accordance with RFC 3325 (http://www.ietf.org/RFC/rfc3325.txt). This method of evaluating phone numbers is used if the number presentation is guaranteed by security mechanisms.
Personalizado: If the Predeterminado and Tal y como se describe en RFC3325 options are unsuccessful, you can also define your own method of evaluating a phone number within certain limits.
Network Provided Number Network-verified incoming phone number: The sequence entered in the invite is searched for the phone number which can be analysed.
User Provided Number Non-verified incoming phone number (corresponds to CLIP no screening): The sequence entered in the invite is searched for the phone number which can be analysed. Optional: is not transferred by all providers.
User-provided name: Caller's name in plain text. Optional: is not transferred by all providers.
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Formato del número llamado
: Format, in which the provider requires the phone number of the called party number to transfer the call. |
Note: The telephone automatically converts called phone numbers into this format when it makes calls via the provider. If you dial a number without a prefix, the local area code therefore applies (Ajustes > Cuentas > Prefijo de país y local).
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Con prefijos de país y local (0044 5306): The called phone number is sent to the provider along with its entire country code (e.g. 0044).
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Con prefijos de país y local (+44 5306): The called phone number is sent to the provider with a plus sign (+) and the entire country code (e.g. +44).
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Con prefijos de país y local (44 5306): The called phone number is sent to the provider with a shortened country code (without international prefix (number), e.g. 44).
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Con prefijo (05306): The called phone number is sent to the provider without a country code.
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Unchanged (as dialled): The called phone number is sent to the provider unchanged.
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Formato del número propio
: Format in which your own phone number, that is to be transmitted, is to be supplied to the provider. |
Note: Your own phone number is automatically converted into the selected format. If you have entered your own phone numbers in the telephone without prefixes, the local area code applies to them during conversion (Ajustes > Cuentas > Prefijo de país y local).
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Con prefijos de país y local (0044 5306): Your own phone number is sent to the provider along with its entire country code (e.g. 0044).
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Con prefijos de país y local (+44 5306): Your own phone number is sent to the provider with a plus sign (+) and the entire country code (e.g. +44).
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Con prefijos de país y local (44 5306): Your own phone number is sent to the provider with a shortened country code (without international prefix (number), e.g. 44).
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Con prefijo (05306): Your own phone number is sent to the provider without a country code.
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Solo número de abonado (MSN/número principal+DDI): Only your own connection number (MSN/DDI main number + DDI) is to sent to the provider.
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Solo extensión (DDI): Only your own Direct Dial In (DDI) is sent to the provider.
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Método para ocultar el número
: Calling line identity restriction by the provider with or without display text. |
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No display text: If calling line identity restriction is enabled, no display text is sent to the called phone number by the provider
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Desconocido: If calling line identity restriction is enabled, the display text "anonymous" is sent to the called phone number by the provider.
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User Anonymous: If calling line identity restriction is enabled, the display text "user anonymous" is sent to the called phone number by the provider.
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Tipo de identificación del número
: Range in which the provider expects to receive phone numbers from the telephone. |
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In the display text: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "Display text" field.
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En el nombre de usuario: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "Username" field.
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Según RFC3325 con P-Asserted-Identity: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "P-Asserted-Identity" field.
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Según RFC3325 con P-Preferred-Identity: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "P-Preferred-Identity" field.
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Personalizar: Uses user-defined settings for number presentation.
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Notes: The Personalizar option requires additional settings:
Note: Some providers already have fixed information for some call types.