Preferences
Proceed as follows:
•Separately for VoIP providers
•Using the Configuration Manager (Réseaux publics > Voice over IP (VoIP page) > Fournisseur > Configurer)
Note:You can obtain the data for the settings either directly from the VoIP provider or from suitable lists in the Internet.Requirements:
•For entering a URL: configured DNS server
•For using a STUN server: IP operating mode IPv4
Fournisseur
Select the provider to be displayed/configured.
Serveur STUN
Adresse IP ou URL | Port
Intervalle pour requête du serveur STUN
Requirements:
•IP operating mode IPv4
•NAT traversalEnabled with use of STUN
A STUN server provides information to subscribers on a private network on request. This information includes the IP address and port outside their private network that is viewed as the source of their data. This information is entered in the requests, instead of the actual private IP address/port.
Note: If you encounter problems with unilateral call connections, enabling the RTP port might help.Note: The Synthèses > Ports page shows an overview of the PBX ports (incoming and outgoing).
Caution: Every time port forwarding is performed, there is a security risk.•For this reason we recommend you use this functionality as little as possible.
Fonctionnement du sous-système
Like any other PBX for which the VoIP provider has been configured, the PBX can be used as a sub-system.
Note: During sub-system operation, single-digit dialling may cause problems. In this case, en-bloc dialling is usually enabled.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 VoIP provider when the connection with the call forwarding destination is established.
Note: If external call forwarding is configured over a VoIP 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).Audio is switched through when call starts
Requirements:
•The provider supports Early Media
Audio is switched through to the provider. This is useful if, for example, a callback request has to be conformed with "Yes".
Support Early-Media
During the call phase audio such as the dialling tone, signal sound, announcement or language can be transmitted (EarlyMedia). The following options define usage of EarlyMedia (RFC 3261) in general and the usage of the SIP Header P-Early-Media (RFC 5009).
Whether EarlyMedia or the SIP Header P-Early-Media is supported, is at the provider's discretion. It is possible that the provider allows the negotiation but prevents the transmission of audio.
The following options allow the adjustment of behaviour of the PBX to the one of the provider.
Switched off
During the call phase no audio is transmitted. But the status of the call is signalled, the end device of the caller then generates for example the dialling tone or the busy tone.
Announcements are as well suppressed. No tones are generated for the caller on incoming calls.
Outgoing without P-Early-Media support
During the call phase audio is negotiated and transmitted for outgoing calls.
The SIP Header P-Early-Media is not sent or evaluated.
Outgoing with P-Early-Media support (standard)
Audio is negotiated and transmitted for outgoing calls.
The SIP Header P-Early-Media is sent and evaluated.
If no P-Early-Media Header is received, EarlyMedia is used in accordance with RFC 3261.
Outgoing with P-Early-Media and incoming only if P-Early-Media is requested
The SIP Header P-Early-Media is sent and evaluated for outgoing calls.
The SIP Header P-Early-Media is only sent and evaluated for incoming calls, if it has been received. Accordingly, audio is transmitted.
This option is only reasonable for sub-system operation and special accounts.
Outgoing and incoming
There are no limitations regarding the negotiation of EarlyMedia for outgoing and incoming calls, the corresponding headers are considered.
Utiliser paramètre URI (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.Utilisation des en-têtes Mediasec selon le projet IETF
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.Le fournisseur peut passer des appels d’urgence
Requirements:
•the associated VoIP provider must allow emergency calls
It is possible to make an emergency call using this provider and the account assigned to it.
Note: When switched off, this provider does not support emergency calls. When you attempt to dial an emergency number, you will hear the announcement, "This phone does not allow emergency calls. Please use an alternative."Déconnexion en cas de modifications NAT
The VoIP 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.
DNS query persists SIP session (RFC 3263)
Requirements:
•SIP stack type 2 selected
The SIP stack saves a resource record entry each for PTR, SRV and AAAA with that the preceded destination was reached. As long as the DNS server delivers this entry it will be prioritised.
Type de pile SIP
Type 1
Software module for the SIP log processing used previously.
Type 2
New extensive software module for the SIP log processing. This software module allows the support of future features of the VoIP providers.
Adresse IP publique statique
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).
Prendre en charge T.38 pour les fournisseurs Négociation T.38
Requirements:
•external VoIP channel that supports T.38 (optional, see Technical Data in the Instructions)
•VoIP provider support for T.38
•Configured internal fax subscribers
T.38 (procedures for real-time Group 3 facsimile communication over IP networks) enables smooth, extensive fax transmission to take place.
Note: If problems occur when T.38 is switched on, you can change the setting used to negotiate the fax protocol (Négociation T.38).Conforme RFC
Two separate media lines for disabling audio and negotiating the fax protocol.
Variant 1
Only one media line for negotiating the fax protocol (the media line for disabling audio is not used).