Help > Using the web interface > Exchange lines > Providers and Accounts > Provider > Settings
Settings
Provider
Select the provider to be displayed/configured.
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STUN server (IP address or URL)
STUN server set for the provider for use of NAT traversal.
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STUN server | Port
Port of the STUN server.
*Note: If you encounter problems with unilateral call connections, enabling the RTP port might help.
*Note: The page Administration > Network > Ports shows an overview of the PBX ports (incoming and outgoing).
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STUN server | Interval for STUN server query
Determines after what time STUN server reuqests are sent.
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Audio is switched through at call forwarding via 2nd B-channel
The connection between the caller and the PBX is always created when a call is forwarded externally. As a result, the caller will hear all replayed elements (tones, announcements) which are played by the provider when the connection with the call forwarding destination is established.
*Note: If external call forwarding is configured over a provider, the call tone may not be replayed for the caller (the connection to the call forwarding destination number will therefore be switched through in a "surprising" way).
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Switch audio on call start
Audio is switched through to the provider provided that the provider supports Early Media. This is useful if, for example, a callback request has to be confirmed with "Yes".
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EarlyMedia Support
During the call phase, audio such as call tone, signal tone, announcement or language can be transmitted (EarlyMedia). The following options define the use of EarlyMedia (RFC 3261) in general and the use of the SIP header P-Early Media (RFC 5009).
If EarlyMedia or the SIP header P-Early Media is supported, is up to the provider's discretion. It is possible that the provider allows the negotiation but suppresses the audio transmission.
The following options allow to adjust the PBX's reaction to the reaction of the provider.
Disabled
No audio is transmitted during the call phase. But the status of the call is signalled and the end device of the caller then generates for example the call tone or the busy tone.
Announcements are still suppressed. No tones are generated for the caller on incoming calls.
Outbound without P-Early Media support
During the call phase audio is negotiated and transmitted on outgoing calls.
The SIP header P-Early Media is not sent or evaluated.
Outbound with P-Early Media support (standard)
Audio is negotiated and transmitted on outgoing calls.
The SIP header P-Early Media is sent and evaluated.
If no P-Early Media header is received, EarlyMedia is used according to RFC 3261.
Outbound with P-Early Media and incoming only if P-Early Media is requested
The SIP header P-Early Media is sent and evaluated on outgoing calls.
On outgoing calls the SIP header P-Early Media is only sent and evaluated if it was received. Audio is transmitted.
This option only makes sense for sub-system operation and special accounts.
Outgoing and incoming
On outgoing and incoming calls, there is no limitation on negotiating EarlyMedia, the appropriate headers are considered.
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Use URI parameter (user)
Transfers the "User=Phone" parameter to the SIP header.
*Note: You may need to switch off this function if there are problems establishing the connection with a particular provider.
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Use Mediasec header in acc. with IETF Draft
Transfers the SIP header in accordance with IETF draft "draft-dawes-sipcore-mediasec-parameter".
*Note: Some providers require this optional SIP header for establishing and running encrypted VoIP connections.
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The provider has emergency call ability
It is possible to make an emergency call using this provider and the account assigned to it.
*Note: When switched off, this provider cannot be used for emergency calls. When you attempt to dial an emergency number, you will hear the announcement, "This telephone does not allow emergency calls. Please use an alternative."
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Deregister, if there are NAT changes
The provider performs the deregistration and then the reregistration.
Some providers require this to take place every time a new public IP address has been assigned.
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DNS query outlasts SIP session (RFC 3263)
The SIP stack saves an entry for each information unit (PTR, SRV and AAAA) with that the preceded destination was reached. As long as the DNS server delivers this entry, it will be prioritised.
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Static public IP address
Some providers provide static public IP addresses and require them for call connection in the VIA header, the Contact header and in the SDP for the audio stream. Enter the provided IP address in the adjacent field and enable the option for the transmission (see previous options).
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Support T.38 for providers
T.38 enables smooth, extensive fax transmission to take place via set internal fax machines.
*Note: If problems occur when T.38 is switched on, you can change the setting used to negotiate the fax protocol (T.38 negotiation).
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