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Using the Configuration Manager (page Settings > Providers & PBXs > Configure)
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int Auerswald PBX sub system IPv4 * profile not selected
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Public exchange (Internet VoIP provider): Selects VoIP provider as the type.
PBX: Selects PBX as the type.
Default network: Default network is used.
VPN: VPN is used.
RegistrarImportant: 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 SIP port set up in the provider's/PBX's configuration.
Registrar port (1...65535) 
Registration interval (min.) (1...60)
Outbound proxyDeactivated: Disables the outbound proxy.
Received automatically: Automatically determines an outbound proxy.
Received manually: Enables you to enter an outbound proxy.
If the Received manually option has been selected, additonal entries must be made:
Outbound Proxy port (1...65535)
Note: You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
SIP NAT traversal|
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STUN server set up for the provider/PBX, if the Enabled with 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.
Enabled : 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.Disabled : 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.Enabled with STUN : NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a STUN server under Settings.
: NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a STUN server under Settings.Note: If Enabled 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 Enabled with 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.
Disabled : 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.Enabled with 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 NAT keep alive interval (sec.) 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 IP/URL STUN server port (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.
SIP port (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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Default SIP server: Server for the normal system telephone operation.
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Broadsoft server: 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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Import
DeleteSIPS: Encryption of SIPS via TLS.
SIP/TLS: Encryption of SIP via TLS.
SIP transport protocolImportant: 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.
Outband, with local response in acc. with 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.Outband, in acc. with 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.Best available quality : 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.Best possible compression : 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.Force 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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Jitter buffer (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.
Disabled: 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.
Preferred: 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.
Mandatory: 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.
Use of public exchange subscriber numbers|
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Set type PBX
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Selected type PBX (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 Public exchange (Internet VoIP provider) under Settings > Providers & PBXs > Configure > Provider > Type some keypad sequencens and the exchange line access number are not available.
Pick-up: 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.
InterCom announcement: 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.
InterCom hands-free calling: The keypad sequence is required to enable Intercom hands-free calling at a destination which does not support automatic call acceptance.
Alarm confirmation: The keypad sequence is required to acknowledge an alarm call with the keypad.
Waiting loop: For moving into the waiting loop.
Standard: 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.
As described in 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.
User-defined: If the Default and As described in 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 of called party number
 : 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 (Settings > Accounts > Country and area code).
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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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With country code and area code (+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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With country code and area code (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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With area code (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 of own number
: 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 (Settings > Accounts > Country and area code).
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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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With country code and area code (+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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With country code and area code (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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With area code (05306): Your own phone number is sent to the provider without a country code.
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Only port number (MSN / DDI + main number): Only your own connection number (MSN/DDI main number + DDI) is to sent to the provider.
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dial (DDI): Only your own Direct Dial In (DDI) is sent to the provider.
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Method for calling line identity restriction
: 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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Anonymous: 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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Manner of number presentation
: 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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In the username: 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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As described in RFC3325 with 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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As described in RFC3325 with 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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Custom: Uses user-defined settings for number presentation.
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Notes: The Custom option requires additional settings:
Note: Some providers already have fixed information for some call types.