Settings Configuration Manager Providers and PBXs

: 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.

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.

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

Requirements:

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Set DNS server

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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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If the Activado option has been selected: the Registrador enabled for the provider/PBX

Important: Every port opening on the NAT router is a security risk. Perform the appropriate security measures.

Desactivado

Activado con uso de STUN

: NAT traversal is performed by the provider/PBX. To do this, you must also specify a STUN server.

NAT-Keep-Alive

Keep alive maintains the connection between client and server by sending NAT keep alive packets which prevent the interruption of the connection. An intveral for the sending of those NAT keep alive packets must be determined. This interval indicates after how many seconds NAT keep alive packets are send to maintain the NAT mapping in the firewall.

NAT keep alive is enabled and an additional entry must be made:

NAT keep alive interval (sec.) (15...255)

Enter NAT keep alive interval.

Note: 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).

STUN server

A STUN server provides subscribers on a private network information on request. This information includes the IP address and port outside their private network that is considered as the source of their data. This information is entered in the requests, instead of the actual private IP address/port.

IP/URL del servidor STUN

Enter STUN server IP/URL.

Puerto del servidor STUN (1...65535)

Enter STUN server port.

STUN server query interval (min) (1...60)

Enter interval.

Requirements:

•

Set DNS server

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.

SIP

Puerto SIP (1...65535)

The SIP port is a port on the local system that is used as a starting point for the SIP transfer.

Notes:

The SIP port must be different for each provider/PBX.

You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.

Enter SIP port.

SIP-Session-Timer

The SIP session timer regularly verifies if the call connection still exists. (The telephone/PBX cannot recognize if the provider interrupts the call connection).

Intervalo para SIP-Session-Timer (min) (5...6)

Indicates after how many minutes the SIP session timer should check the call connection.

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.

Tipo de servidor

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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.

Session Initiation Protocol Security (SIPS)

Requirements:

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SIP account: A certificate, saved on the hard disk

Important: If you are using this on the PBX, the PBX specifies the encryption method. The PBX overwrites manual settings.

Overhearing of VoIP conversations will be prevented, as these connections are encrypted. The connection setup and termination, and also signalling, are encrypted with SIPS. The call data is encrypted with SRTP.

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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Set PBX Account: Huella dactilar para certificado SIPS

. Enter fingerprint.

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Set SIP account: Importar certificado

. Import certifcate from the hard disk.

Importación

Borrar

SIPS: Encryption of SIPS via TLS.

SIP/TLS: Encryption of SIP via TLS.

Desactivar comprobación de nombres de host

No verification whether the certificate belongs to the domain/IP.

Protocolo de transporte SIP

UDP (User Datagram Protocol) is used to send data packet over non-secure communication lines without a connection. This means that successful transmission is dependent on the application and is therefore not always guaranteed. UDP itself does not verify whether data has been transmitted successfully. When a UDP packet is sent, the sender cannot assume that the packet will indeed arrive at the recipient. This particular protocol needs only a small amount of additional information, and results in a better data throughput rate in a well-functioning network, e.g., on a LAN. UDP is used, e.g., for the DNS (Domain Name Server).

TCP (Transmission Control Protocol) is a transport protocol that segments data into packets up to a specified size and reliably sends these individual data packets in the correct sequence to the recipient address. In this process, every data packet sent must be resent until it has been confirmed as arrived. In order to make sure this happens, a large amount of information is sent along with the actual payload data. Most Internet services are implemented with TCP, e.g., HTTP (WWW), SMTP/POPS (e-mail), etc.

Important: If encryption is enabled by SIPS, the TCP transport protocol is used. Manual settings are overwritten.

UDP: UDP is used.

TCP: TCP is used.

RTP

Señalización DTMF

At DTMF signalling, a sound of two overlapped sound frequencies is assigned to each digit of the keypad. Here, you must select how the DTMF signalling is to be transmitted. This is critically depened on the provider and the transmission path.

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.

Inband, DTMF signals sent through the audio channel

: 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.

Codecs

A codec is a method that encodes (digitises) analogue voice data for transmission and again decodes again, meaning converts into back into voice. There are various codecs that feature different voice data compression rates thereby require different band widths for data transmission. The equality of VoIP calls is dependent on the codec used.

Notes:

If a codec with high bandwidth (e.g. G.711) produces too much interference in the call quality, the connection bandwidth might not be sufficient. If these types of disruptions happen frequently, we recommend you select codecs with a lower band width.

Not every codec is supported by every provider.

Máxima calidad

: 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.

Forzar G711

: Codec G.711 is required.

Música de espera

Requirements:

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int Auerswald PBX sub system IPv4 * profile not selected

When putting a caller on hold, music on hold will be played.

Búfer de jitter (ms) (40...160)

The size of the jitter buffer shows how many RTP packets can be buffered, to bridge or compensate disruptions.

Jitter buffer size entry in milliseconds (duration of the audio signal).

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.

SRTP

To prevent anyone from eavesdropping VoIP conversations, you can encrypt these connections. The connection setup and termination, and also signalling, are encrypted with SIPS. The call data is encrypted with SRTP.

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.

Numbers

Utilización de números de línea externa

Public exchange subscriber numbers must be used for PBXs which allow you to dial external telephone numbers without preceding exchange line access numbers or characters (e. g. 0 or **). Such PBXs automatically detect whether exchange line access is required or not.

Requirements:

•

No operation on the PBX

•

Set type Centralita

Call acceptance rules for internal and external will be applied when a call comes in. In addition, you can forward an incoming call to an external telephone number without having to dial the exchange line access number.

Número de acceso a línea externa

Requirements:

•

Selected type Centralita (page Settings > Providers & PBXs > Provider)

Note:

After you dial the exchange line access number, the external dial tone indicates that an external line is free. The external number can then be dialled.

You can also request an exchange line from the telephony app. Tap on

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.

The exchange line access number of the PBX is „0“ in the factory settings but can be changed in some PBXs.

Enter the exchange line access number. When used as a system telephone the exchange line access number is automatically transferred from the PBX.

Secuencia de keypad

Keypad sequences enable you to control features by entering characters and number sequences via the phone keys, e.g. to make pick-ups or InterCom announcements.

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.

Enter keypad sequence

: The following keypad sequences for PBX operation are 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.

Análisis de mensajes SIP entrantes

Depending on the SIP provider, the phone number of the caller can be in different locations in the SIP message. The following options and settings must be selected and made to determine at which location of the SIP message the phone number shall be searched.

Two different phone numbers can be transmitted: The actual phone number of the caller (screened number/network provided number) and a number self-selected by the caller (unscreened number/user provided number). If a network provided number is extracted from the SIP message, functions such as Call Through can be used.

Tipo de análisis

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.

Notes:

The phone number to be evaluated should preferably be canonical (+445306.... or 00445306…). Otherwise it is not used for evaluation.

The designations used for the settings match the designations used in Wireshark.

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.

Internacionalizar números desconocidos

Phone numbers which are not transferred in the usual format: (e. g. +445306… or 00445306…) are converted into this format.

Identificación del número (saliente)

You need to input the following information to ensure that the telephone transfers the data (e.g. the called phone number, its own phone number) to the provider in the correct format:

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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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With country and area code (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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With country and area code (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).

–

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.

–

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.

Notes: The Personalizar option requires additional settings:

From

Settings for setting up the "From" header.

P-Asserted-Identity

Settings for setting up the P-Asserted-Identity" header.

P-Preferred-Identity

Settings for setting up the "P-Preferred-Identity" header.

Alert-info

Definición de tipos de llamada por información de alerta

A specific ringtone can be assigned to the different call types of a SIP account. The required information is transmitted by the SIP provider.