<div dir="ltr">Hey guys and girls,<br><br>I need some help setting up a SIP trunk to a CME router from a CUCM6 server.<br><br>I have followed the following guide (<a href="http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration">http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration</a>) to configure the CUCM6 and as far as I can see it's working as I'm able to make calls the remote site.<br>
<br>The configuration on the remote router is as follows:<br><br>dial-peer voice 11 voip<br> description Head Office<br> translation-profile outgoing strip-to-3digext<br> preference 5<br> destination-pattern [1-4].<br> session protocol sipv2<br>
session target ipv4:<a href="http://10.103.10.5">10.103.10.5</a><br> dtmf-relay sip-notify<br> codec g711ulaw<br> no vad<br><br>The problem is that the remote site can not call Head Office, any ideas?<br><br>Thanks<br></div>