Hi<br><br>This works fine in extensions.conf:<br><br>exten => _0X./100,1,Dial(SIP/${EXTEN}@sipout-a)<br>exten => _0X./200,1,Dial(SIP/${EXTEN}@sipout-a)<br><br>This will just use different SIP channels for different Caller ID's.
<br>If I write the same to a realtime table, Asterisk always uses sipout-a, no matter what Caller ID is used. <br><br>