SIP-Trunk Details

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 ForwardDTMF 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
  • 89.202.102.86
  • 89.202.102.83
  • 89.202.102.84

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.

 

Asterisk Configuration

Asterisk Configuration:

Config file: sip.conf

[TENIOS-TRUNK]
type=friend
host=200000.tenios.com
fromdomain=200000.tenios.com
insecure=port,invite
username=tenios
secret=mysecretpassword
nat=yes
canreinvite=no
qualify=no
cancallforward=yes
context=tenios
language=de
dtmfmode=rfc2833 ;its a test
sendrpid=yes
callevents=yes
disallow=all
allow=alaw
allow=ulaw

 

The Parameter host, has the format <customer number>.tevox.com. If you have the customer number 1000000, please enter here 1000000.tevox.com . The parameters username and secret, must match the values in the TEVOX SIP trunk configuration. To route calls through the configured SIP Trunk to the TEVOX platform, you must add the following entry in the configuration file extensions.conf of your Asterisk server:

 

Config file: extensions.conf

exten => 3333,1,Dial(SIP/022155400300@TEVOX-TRUNK,60,tr)

 

Freeswitch Configuration

Freeswitch-Configuration

First, add a gateway to your SIP profile (/usr/local/freeswitch/conf/sip_profiles/external.xml). The parameter proxy, has the format <customerno>.tevox.com. If you have the customer number 1000000, please enter 1000000.tevox.com ein here. The parameters username and password, must match the values in the TENIOS SIP-Trunk configuration:

<gateway name="mytestgateway">
    <param name="username" value="myUsername"/>
    <param name="password" value="myPassword"/>
    <param name="proxy" value="200002.tevox.com:5090"/>
    <param name="register" value="true"/>
 </gateway>

 

To route calls through the configured SIP-Trunk to the TENIOS platform, you must bridge calls over the configured gateway:

<extension name="TEVOX mytestGateway">
     <condition>
       <action application="bridge" data="{sip_h_P-Called-Party-ID=${destination_number}}sofia/gateway/mytestgateway/${destination_number}"/>
     </condition>
   </extension>

 

 

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.

Register trunkIP AddressRegister Trunk Outbound

register trunk

 

sip trunk ip adress

 

sip trunk outbound

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: 

Register Trunk
The other endpoint registers with this trunk on TENIOS.
IP-Address
The other endpoint is defined by IP address. For those kind of trunks only forwarding is possible. TENIOS can send calls to the configured IP address, but it is not possible to send calls to TENIOS on this trunk.
Register Trunk Outbound
Pleaser enter SIP address of the authentication proxy.
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. 
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