Hello, Some background before i ask the question. I am attempting to implement?a SIP?trunk?between an askerisk an?a Mitel 5000 system.? The mitel is giving me 404 errors when I send a call over to it even though the call desination is valid. The mitel also reports an error saying cp_dest_id is NULL.??(Call?Processor destination?ID is what?I assume that means).? The Mitel has an incomming routing mechanism that says this in the help: "....the system will look for the dialed number provided by DID (DDI in Europe) or DNIS."? In reference to incomming trunks. ? So, I am thinking there is no DID or DNIS information being send accross.? I caputed packets and I did not see a DID or DNIS field in the headers.? I am esentially trying to make the asterisk system look like a SIP provider (bandwidth.com for instance) with DIDs.? I can call from the Mitel to the Asterisk?but i just cannot get the call to come back the other direction.? ? What does it take to send information like this (DID and or DNIS)?over?a SIP trunk?? Can it be done and what should the headers, etc actually look like when looking at the packets??I basically want the asterisk to talk like a SIP DID and Trunk service?provider. ? -Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110413/b61354a1/attachment.htm>