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[[File:nuovoDominio.png|miniatura]]
[[File:nuovoDominio.png|miniatura]]
For every VoIP domain, you can set up any combination of VoIP terminations and trunks.  
For every VoIP domain, you can set up any combination of VoIP terminations and trunks.  
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For outbound lines, it is possible to use each as an independent line. For inbound lines, all calls from that domain will be handled by the "VoIP domain" entity, which contains all the routing rules related to the numbering of the trunks and terminations that belong to it.
For outbound lines, it is possible to use each as an independent line. For inbound lines, all calls from that domain will be handled by the "VoIP domain" entity, which contains all the routing rules related to the numbering of the trunks and terminations that belong to it.


Nella tabella seguente sono illustrati i parametri che è possibile configurare per un dominio VoIP.
The following table lists the configurable parameters for a VoIP domain.


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{| class="wikitable"
{| class="wikitable"
|-
|-
! <div style = 'text-align: left> '''Parametro''' !! <div style = 'text-align: left> '''Descrizione''' !! <div style = 'text-align: left> '''Valore'''
! <div style = 'text-align: left> '''Parameter''' !! <div style = 'text-align: left> '''Description''' !! <div style = 'text-align: left> '''Value'''
|-
|-
|colspan="3"| <div style = 'text-align: center; color: transparent> -
|colspan="3"| <div style = 'text-align: center; color: transparent> -
|-
|-
| Abilitato || Abilitare o disabilitare un dominio VoIP senza perderne la configurazione || On / Off
| Enabled || Lets you disable a VoIP domain without losing its configuration. || On / Off
|-
|-
| Nome || Nome mnemonico assegnato al dominio VoIP || Alfa-numerico
| Name || Mnemonic name assigned to the VoIP domain. || Alphanumeric
|-
|-
!colspan="3"| <div style = 'text-align: left> '''''Impostazioni SIP'''''
!colspan="3"| <div style = 'text-align: left> '''''SIP settings'''''
|-
|-
| Indirizzo IP del server || L'indirizzo IP oppure l'hostname del dominio VoIP || Hostname / Indirizzo IP
| Server IP address || The address or hostname of the VoIP domain. || Hostname / IP address
|-
|-
| Porta del server ||La porta utilizzata dal server del dominio VoIP. Nel caso in cui per il server sia indicato l'hotstname può essere omessa dal momento che KalliopePBX ricava automaticamente la porta facendo una query [https://tools.ietf.org/html/rfc2782 SRV]. Se inserita KalliopePBX userà comunque questa, anche in caso di query SRV. || Numerico
| Server port || The port used by the VoIP domain. This can be omitted if the hostname is specified, as KalliopePBX will automatically acquire the port with an [https://tools.ietf.org/html/rfc2782 SRV] query. If specified anyway, KalliopePBX will use it regardless of the SRV query. || Numeric
|-
|-
| Abilita porta non sicura || Permette di abilitare il riconoscimento del peer di origine della chiamata sulla base del solo indirizzo IP, ignorando la porta sorgente. Questa impostazione risulta utile quando tra il KalliopePBX ed il gateway sia presente un firewall che altera la porta sorgente dei messaggi SIP, impedendo il corretto ingresso delle chiamate sul sistema. || Si / No
| Enable insecure port || Let you enable recognition of the origin peer of the base call based only on IP address, ignoring the source port. This setting is useful when there is a firewall between KalliopePBX and the gateway that alters the source port of SIP messages, preventing calls from correctly entering the system. || Yes / No
|-
|-
| Abilita SRTP || Abilita il supporto all'[https://tools.ietf.org/html/rfc3711 SRTP] per questo dominio VoIP. || Si / No
| Enable SRTP || Enables [https://tools.ietf.org/html/rfc3711 SRTP] support for this VoIP domain. || Yes / No
|-
|-
!colspan="3"| <div style = 'text-align: left> '''''Impostazioni di trasporto'''''
!colspan="3"| <div style = 'text-align: left> '''''Transport settings'''''
|-
|-
| Tipo di trasporto || Permette di scegliere per questo dominio se utilizzare SIP con trasporto UDP/TCP/TLS oppure SIP con trasporto Web Socket/Web Socket Sicuro || SIP / WebRTC
| Transport type || Lets you choose between SIP with UDP/TCP/TLS transport or SIP with WebSocket/Secure WebSocket transport for this domain. || SIP / WebRTC
|-
|-
| Trasporto preferito || Menu a tendina che permette di indicare quale sia il metodo di trasporto preferito tra quelli abilitati. Qualora venda selezionato un metodo di trasporto disabilitato, quest'ultimo verrà immediatamente reso abilitato. || UDP / TCP / TLS / WS / WSS
| Preferred transport || Drop-down menu that lets you choose the preferred transport among those enabled. Selecting a disabled transport will automatically enable it. || UDP / TCP / TLS / WS / WSS
|-
|-
| Abilita trasporto UDP || Abilita il trasporto su protocollo [https://www.ietf.org/rfc/rfc768.txt UDP] non cifrato. Disponibile solo se alla voce "Tipo di trasporto" si è scelto SIP. || Si / No
| Enable UDP transport || Enable unencrypted [https://www.ietf.org/rfc/rfc768.txt UDP] transport. Only available if "Transport type" is set to SIP. || Yes / No
|-
|-
| Abilita trasporto TCP|| Abilita il trasporto su protocollo [https://tools.ietf.org/html/rfc793 TCP] non cifrato. Disponibile solo se alla voce "Tipo di trasporto" si è scelto SIP. || Si / No
| Enable TCP transport || Enable unencrypted [https://tools.ietf.org/html/rfc793 TCP] transport. Only available if "Transport type" is set to SIP. || Yes / No
|-
|-
| Abilita trasporto TLS|| Abilita il trasporto su protocollo cifrato [http://www.rfc-base.org/txt/rfc-5246.txt TLS]. Disponibile solo se alla voce "Tipo di trasporto" si è scelto SIP. || Si / No
| Enable TLS transport || Enable encrypted [http://www.rfc-base.org/txt/rfc-5246.txt TLS] transport. Only available if "Transport type" is set to SIP. || Yes / No
|-
|-
| Abilita trasporto Web Socket|| Abilita il trasporto su protocollo [https://tools.ietf.org/html/rfc6455 WS (Web Socket)] non cifrato. Disponibile solo se alla voce "Tipo di trasporto" si è scelto Web Socket. || Si / No
| Enable WebSocket transport || Enable unencrypted [https://tools.ietf.org/html/rfc6455 WS (WebSocket)] transport. Only available if "Transport type" is set to WebSocket. || Yes / No
|-
|-
| Abilita trasporto Web Socket Sicuro|| Abilita il trasporto su protocollo cifrato [https://tools.ietf.org/html/rfc6455 WSS (Web Socket Sicuro)]. Disponibile solo se alla voce "Tipo di trasporto" si è scelto Web Socket. || Si / No
| Enable WebSocket transport || Enable encrypted [https://tools.ietf.org/html/rfc6455 WSS (Secure WebSocket)] transport. Only available if "Transport type" is set to WebSocket. || Yes / No
|-
|-
!colspan="3"| <div style = 'text-align: left> '''''Impostazioni avanzate'''''
!colspan="3"| <div style = 'text-align: left> '''''Advanced settings'''''
|-
|-
| Estrazione numero dall'header "To" || Permette di estrarre il numero chiamato dal "To:" URI invece che dal "Request-URI". Richiesto da alcuni provider VoIP || Si / No
| Extract number from "To:" header || Extract the called number from the "To:" header instead of the "Request-URI". Required by some VoIP providers. || Yes / No
|-
|-
| Modalità DTMF || Definisce la modalità di invio dei toni DTMF verso questo gateway, tra quelle previste (RFC 2833, SIP Info, in audio). Di default è impostato alla modalità definita come predefinita per il sistema. || RFC 2833 / SIP Info / In audio
| DTMF mode || Choose how DTMF tones are sent to this gateway, among the modes provided (RFC 2833, SIP Info, in audio). By default, this will be set to the predefined system mode. || RFC 2833 / SIP Info / In audio
|-
|-
| Abilita accettazione RPID || Indica se l’header Remote Party ID (RPID) o P-Asserted-Identity (PAI) debba essere rispettato per le chiamate provenienti da quel dominio || Default di sistema / Abilitato / Disabilitato
| Respect RPID || Choose whether or not the Remote Party ID (RPID) or P-Asserted-Identity (PAI) header must be respected for calls from this domain. || System default / Enabled / Disabled
|-
|-
| Modalità invio RPID || Permette di scegliere se indicare l’identità dell’utente che effettua la chiamata in uscita verso quel dominio tramite l'header P-Asserted-Identity (PAI) oppure tramite il Remote-Part-ID (RPID), al fine dell’aggiornamento della presentazione dell’informazione della linea connessa (COLP) || Default di sistema / Disabilitato / Remote-Part-ID / P-Asserted-Identity
| RPID sending mode || Choose whether to show the identity of the calling party through P-Asserted-Identity (PAI) or Remote-Part-ID (RPID), in order to update the connected line identification presentation (COLP). || System default / Disabled / Remote-Part-ID / P-Asserted-Identity
|-
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|}
|}
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Versione attuale delle 21:58, 25 set 2022

Altre lingue:

Return to AdminGuide:BasicConcepts

NuovoDominio.png

For every VoIP domain, you can set up any combination of VoIP terminations and trunks.

For outbound lines, it is possible to use each as an independent line. For inbound lines, all calls from that domain will be handled by the "VoIP domain" entity, which contains all the routing rules related to the numbering of the trunks and terminations that belong to it.

The following table lists the configurable parameters for a VoIP domain.

Parameter
Description
Value
-
Enabled Lets you disable a VoIP domain without losing its configuration. On / Off
Name Mnemonic name assigned to the VoIP domain. Alphanumeric
SIP settings
Server IP address The address or hostname of the VoIP domain. Hostname / IP address
Server port The port used by the VoIP domain. This can be omitted if the hostname is specified, as KalliopePBX will automatically acquire the port with an SRV query. If specified anyway, KalliopePBX will use it regardless of the SRV query. Numeric
Enable insecure port Let you enable recognition of the origin peer of the base call based only on IP address, ignoring the source port. This setting is useful when there is a firewall between KalliopePBX and the gateway that alters the source port of SIP messages, preventing calls from correctly entering the system. Yes / No
Enable SRTP Enables SRTP support for this VoIP domain. Yes / No
Transport settings
Transport type Lets you choose between SIP with UDP/TCP/TLS transport or SIP with WebSocket/Secure WebSocket transport for this domain. SIP / WebRTC
Preferred transport Drop-down menu that lets you choose the preferred transport among those enabled. Selecting a disabled transport will automatically enable it. UDP / TCP / TLS / WS / WSS
Enable UDP transport Enable unencrypted UDP transport. Only available if "Transport type" is set to SIP. Yes / No
Enable TCP transport Enable unencrypted TCP transport. Only available if "Transport type" is set to SIP. Yes / No
Enable TLS transport Enable encrypted TLS transport. Only available if "Transport type" is set to SIP. Yes / No
Enable WebSocket transport Enable unencrypted WS (WebSocket) transport. Only available if "Transport type" is set to WebSocket. Yes / No
Enable WebSocket transport Enable encrypted WSS (Secure WebSocket) transport. Only available if "Transport type" is set to WebSocket. Yes / No
Advanced settings
Extract number from "To:" header Extract the called number from the "To:" header instead of the "Request-URI". Required by some VoIP providers. Yes / No
DTMF mode Choose how DTMF tones are sent to this gateway, among the modes provided (RFC 2833, SIP Info, in audio). By default, this will be set to the predefined system mode. RFC 2833 / SIP Info / In audio
Respect RPID Choose whether or not the Remote Party ID (RPID) or P-Asserted-Identity (PAI) header must be respected for calls from this domain. System default / Enabled / Disabled
RPID sending mode Choose whether to show the identity of the calling party through P-Asserted-Identity (PAI) or Remote-Part-ID (RPID), in order to update the connected line identification presentation (COLP). System default / Disabled / Remote-Part-ID / P-Asserted-Identity
Estratto da "https://www.kalliope.com/wiki/index.php?title=AdminGuide:BasicConcepts:Outbound_lines:Dominio_VoIP/en&oldid=17938"