Add a VoIP domain or edit an existing one
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
Parameter |
Description |
Value |
|---|---|---|
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
Parameter |
Description |
Value |
|---|---|---|
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 (Web Socket) transport. Only available if “Transport type” is set to WebSocket |
Yes / No |
Enable WebSocket transport |
Enable encrypted WSS (Secure Web Socket) transport. Only available if “Transport type” is set to WebSocket |
Yes / No |
Advanced settings
Parameter |
Description |
Value |
|---|---|---|
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 |
Audio codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of audio Codecs |
PCM a-law / G722 (HD Audio) / G.726 / G.729 / GSM / Opus / PCM u-law |
Video codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of video Codecs |
H.261 / H.263 / H.263+ / H.264 / VP8 |
Caller and called identifier mapping rules (incoming calls)
Parameter |
Description |
Value |
|---|---|---|
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Add a physical gateway or edit an existing one
The following table lists the configurable parameters for a gateway.
Parameter |
Description |
Value |
|---|---|---|
Enabled |
Lets you disable a gateway without losing its configuration |
On / Off |
Name |
Mnemonic name assigned to the gateway |
Alphanumeric |
SIP settings
Parameter |
Description |
Value |
|---|---|---|
Identifier |
Gateway identifier |
Alphanumeric |
Server IP address |
The address or hostname of the gateway |
Hostname / IP address |
Server port |
The port used by the gateway. 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 inbound registration |
Yes / No |
|
Do not request authentication for calls from this peer |
Yes / No |
Transport settings
Parameter |
Description |
Value |
|---|---|---|
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 Web RTC transport |
Enable Web RTC transport protocol |
Yes / No |
Advanced settings
Parameter |
Description |
Value |
|---|---|---|
Simultaneous call limit |
Number of simultaneous calls allowed |
Numeric |
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 |
COLP sending mode |
System default / Disabled / Remote-Part-ID / P-Asserted-Identity |
|
COLP acceptance mode |
System default / Disabled / Enabled |
Audio codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of audio Codecs |
PCM a-law / G722 (HD Audio) / G.726 / G.729 / GSM / Opus / PCM u-law |
Video codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of video Codecs |
H.261 / H.263 / H.263+ / H.264 / VP8 |
Caller and called identifier mapping rules (incoming calls)
Parameter |
Description |
Value |
|---|---|---|
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller and called identifier mapping rules (outgoing calls)
Parameter |
Description |
Value |
|---|---|---|
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Telephone numbers belonging to the line
Parameter |
Description |
Value |
|---|---|---|
Phone numbers |
Numeric |
Add a VoIP termination or edit an existing one
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 termination.
Parameter |
Description |
Value |
|---|---|---|
Enabled |
Lets you disable a VoIP termination without losing its configuration. |
On / Off |
Name |
Mnemonic name assigned to the VoIP termination |
Alphanumeric |
SIP settings
Parameter |
Description |
Value |
|---|---|---|
VoIP domain |
Select the hostname of the VoIP domain to which you wish to associate the termination |
Hostname |
Source user |
Name of the source |
Alphanumeric |
Username |
Username with which the trunk is accessed |
Alphanumeric |
Secret |
Alphanumeric |
Transport settings
Parameter |
Description |
Value |
|---|---|---|
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 Web RTC transport |
Enable Web RTC transport protocol |
Yes / No |
Advanced settings
Parameter |
Description |
Value |
|---|---|---|
Simultaneous call limit |
Number of simultaneous calls allowed |
Numeric |
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 |
COLP sending mode |
System default / Disabled / Remote-Part-ID / P-Asserted-Identity |
|
COLP acceptance mode |
System default / Disabled / Enabled |
Video codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of video Codecs |
H.261 / H.263 / H.263+ / H.264 / VP8 |
Header SIP
Parameter |
Description |
Value |
|---|---|---|
Set the user part of the From URI as |
Caller number after manipulation / Caller number before manipulation / Authentication username / Custom. |
|
Sets the display name in the From header to the user part of the From URI |
Yes, except to remote extensions / No (keep the original display name) / Yes |
Show available placeholders
Parameter |
Description |
Value |
|---|---|---|
P-Asserted-Identity / P-Preferred-Identity / Remote-Party-ID / Call-Info |
Caller and called identifier mapping rules (outgoing calls)
Parameter |
Description |
Value |
|---|---|---|
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Telephone numbers belonging to the line
Parameter |
Description |
Value |
|---|---|---|
Phone numbers |
Numeric |
Add a VoIP trunk or edit an existing one
For every VoIP domain, you can set up any combination of VoIP terminations and trunks.
The following table lists the configurable parameters for a trunk.
Parameter |
Description |
Value |
|---|---|---|
Enabled |
Lets you disable a trunk without losing its configuration |
On / Off |
Name |
Mnemonic name assigned to the gateway |
Alphanumeric |
SIP settings
Parameter |
Description |
Value |
|---|---|---|
VoIP domain |
Select the hostname of the VoIP domain to which you wish to associate the trunk |
Hostname |
Identifier |
Trunk identifier |
Alphanumeric |
Username |
Username with which the trunk is accessed |
Alphanumeric |
Secret |
Alphanumeric |
|
Registration validity |
Numeric |
|
Source domain (registration) |
Alphanumeric |
|
Source domain |
Alphanumeric |
Transport settings
Parameter |
Description |
Value |
|---|---|---|
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 Web RTC transport |
Enable Web RTC transport protocol |
Yes / No |
Advanced settings
Parameter |
Description |
Value |
|---|---|---|
Simultaneous call limit |
Number of simultaneous calls allowed |
numeric |
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 |
COLP sending mode |
System default / Disabled / Remote-Part-ID / P-Asserted-Identity |
|
COLP acceptance mode |
System default / Disabled / Enabled |
Audio codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of audio Codecs |
PCM a-law / G722 (HD Audio) / G.726 / G.729 / GSM / Opus / PCM u-law |
Video codec
Parameter |
Description |
Value |
|---|---|---|
Allows you to choose the types of video Codecs |
H.261 / H.263 / H.263+ / H.264 / VP8 |
Header SIP
Parameter |
Description |
Value |
|---|---|---|
Set the user part of the From URI as. |
Caller number after manipulation / Caller number before manipulation / Authentication username / Custom. |
|
Sets the display name in the From header to the user part of the From URI. |
Yes, except to remote extensions / No (keep the original display name) / Yes |
Show placeholders available
Parameter |
Description |
Value |
|---|---|---|
P-Asserted-Identity / P-Preferred-Identity / Remote-Party-ID / Call-Info |
Caller and called identifier mapping rules (outgoing calls)
Parameter |
Description |
Value |
|---|---|---|
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Filters |
Any / Exact / Prefix / Range |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Caller Manipulation |
Remove + Pref. |
Numeric |
Remote extensions
Parameter |
Description |
Value |
|---|---|---|
Type of selection |
Exact selection / Range of selection / Prefix selection. |
|
Selection value |
Numeric |
|
Routing class |
