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

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>