A SIP-Trunk is a direct VoIP connection between a communication server in your company and the TENIOS platform. Unlike a connection with a SIP softphone, a SIP trunk can handle multiple calls simultaneously. The maximum number of channels is limited to 30 per SIP trunk. In the SIP trunk configuration, it can be specified if the endpoint registers with the SIP Trunk. If you have configured a trunk with registration, your communication server has to register with the TENIOS platform. Below you will find some sample configurations which outline how to connect an Asterisk or Freeswitch server to the TENIOS platform via SIP trunking.
Call Forwarding to SIP-Trunks
You can also route incoming calls to SIP-Trunks. The configured SIP-Trunks are available as destinations in the routingplan for the respective blocks Forward, DTMF and Queue. The Call forwarding is done by different servers, e.g. the source IP in a SIP INVITE message can be set to the following IP addresses:
- 80.239.139.194
- 80.239.139.204
- 80.239.139.207
- 80.239.139.208
- 89.202.102.83
- 89.202.102.84
- 89.202.102.85
- 89.202.102.86
Please configure your firewall or the ACL (Access Client List) settings of your telephony switch settings appropriate so that forwarded calls are accepted.
The Configuration Guides below show how to connect Asterisk and Freeswitch with the TENIOS platform via SIP-Trunk.
Configure SIP-Trunk details
On this page you can configure SIP-Trunks. In the table below you find some configuration instructions selecting your type of Trunk.
Field | Description |
Name | The name of the SIP-Trunk. This name is displayed in a routing plan blocks when a call forward to a trunk is configured. |
Type of Trunk | The type of the trunk:
|
Auth proxy | The SIP address of the authentication proxy, e.g., sipprovider.com[:port] or an IP address[:port]. The port is optional. |
Realm | Remote realm e.g.: sipprovider.com or and IP address. INFO: When this trunk type is selected the sip trunk will register |
IP/Domain | The IP or domain name of the remote endpoint. |
Transport protocol | The transport protocol for the SIP trunk can be specified here. This parameter is only required for trunks of type “IP address”. |
Username | The username of the SIP trunk. |
Password | The password of the SIP trunk. |
Always Show Caller ID | Hint: For inbound calls, keep PID and Privacy Headers. |
Clip No Screening | Hint: Clip no Screening does not check if the Callerid is owned by this account. |
Active | A flag if the trunk is active. If the trunk is not active, it is not possible to register with the trunk or send calls to it. |
Prefix manipulation | In this section you can set a prefix to the CLI or DNIS. This is only applied in the direction TEVOX -> Customer PBX. |