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Using the Configuration Manager (page Paramètres > Fournisseurs et IPBX > Configurer)
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int Auerswald PBX sub system IPv4 * profile not selected
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Fournisseur VoIP (internet) Selects VoIP provider as the type.
IPBX Selects PBX as the type.
Réseau par défaut Default network is used.
VPN VPN is used.
RegistreImportant: 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 Port SIP set up in the provider's/PBX's configuration.
Port du registre (1...65535) 
Délai pour l’enregistrement (min) (1...60)
Proxy sortantDésactivé: Disables the outbound proxy.
Affectation automatique: Automatically determines an outbound proxy.
Affectation manuelle: Enables you to enter an outbound proxy.
If the Affectation manuelle option has been selected, additonal entries must be made:
Port proxy sortant (1...65535)
Note: You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
NAT Traversal SIP|
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STUN server set up for the provider/PBX, if the activé avec utilisation 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.
activé : 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.désactivé : 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.activé avec utilisation de STUN : NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a Serveur STUN under Paramètres.
: NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a Serveur STUN under Paramètres.Note: If activé 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 activé avec utilisation 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.
désactivé : 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.activé avec utilisation 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. NAT Keep AliveNote: Some providers lock accounts if the Intervalle 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).
Adresse IP / URL serveur STUN Port du serveur 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.
Port 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.
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Serveur SIP par défaut: Server for the normal system telephone operation.
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Serveur 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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Importation
SupprimerSIPS: Encryption of SIPS via TLS.
SIP/TLS: Encryption of SIP via TLS.
Protocole de transport 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.Note: If your are unsure which DMFT signalling type you must select, you cand find out the right type by trying them out.
Sortant avec tonalité d’acquittement locale, selon 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.Sortant, selon 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.Meilleure qualité possible : 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.Meilleure compression possible : 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.Forcer 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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Tampon de gigue (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.
désactivé: 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.
privilégié: 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.
obligatoire: 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.
Utilisation de numéros de téléphone de la ligne principale|
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Set type IPBX
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Selected type IPBX (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 Fournisseur VoIP (internet) under Paramètres > Fournisseurs et IPBX > Configurer > Fournisseur d’accès > Type some keypad sequencens and the exchange line access number are not available.
Interception d’appel: 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.
Annonce 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.
Fonction mains-libres InterCom: The keypad sequence is required to enable Intercom hands-free calling at a destination which does not support automatic call acceptance.
Acquittement de l’alarme: The keypad sequence is required to acknowledge an alarm call with the keypad.
File d’attente: For moving into the waiting loop.
Par défaut: 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.
Comme décrit dans 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.
personnalisé: If the Par défaut and Comme décrit dans 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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Format du numéro de téléphone composé
 : 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 (Paramètres > Comptes > Indicatif international et indicatif régional).
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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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Avec indicatif international et indicatif régional (+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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Avec indicatif international et indicatif régional (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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Avec indicatif régional (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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Format du numéro de téléphone personnel
: 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 (Paramètres > Comptes > Indicatif international et indicatif régional).
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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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Avec indicatif international et indicatif régional (+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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Avec indicatif international et indicatif régional (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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Avec indicatif régional (05306): Your own phone number is sent to the provider without a country code.
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Uniquement numéro de la ligne (MSN/numéro de téléphone principal+DDI): Only your own connection number (MSN/DDI main number + DDI) is to sent to the provider.
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Uniquement appel direct (DDI): Only your own Direct Dial In (DDI) is sent to the provider.
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Méthode pour la non-présentation du numéro de téléphone
: 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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Inconnu: 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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Type de présentation du numéro de téléphone
: 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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Dans le nom d’utilisateur: 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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Selon RFC3325 avec 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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Selon RFC3325 avec 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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Personnalisé: Uses user-defined settings for number presentation.
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Notes: The Personnalisé option requires additional settings:
Note: Some providers already have fixed information for some call types.