[Serusers] Call Forward Uncontitionally

Hi to all, i have to implement the call transfer service for our offices using SER. We have a PSTN gateway (cisco 3725) that receive calls from PSTN to out VOIP equipments. I have already implemented an interface to let the user to activate the CFU and set it in a database. I made an external script to check this database (made in php). If is the CFU it active it rewrite the uri with the new destination number. Now i have accomplished to all work but i don't know how to make an IF statement based on a result of an esternal script i.e.: if(external_script==1) do the rewriting; else continue with the traversal of the rules. I need to do this in this way becouse after this check i have other rules that manage outgoing calls from VOIP so i need that if the CFU is not active the next rule have to be checked to determine the real destination address. After this i have another problem: If the call transfer is active the source number is leaved as is and the central Phone Switch do not allow to pass that number as source number, so it keep the default number (or anonymous) to originate the call. This becouse being on the client side we cannot generate the calls with all the source we want. I place the example: A) Calling number: 123456 (from PSTN) B) Called Number: 555555 (on VOIP) C) Transferred number : 999999 (on PSTN selected by VOIP user 555555 to transfer the call to) D) Default Number of the PRI : 111111 I am the user B and activate the call transfer to C. A call come from A to B. The SIP proxy receive the call and change the destination from B to C and mantain the source as A. When the Cisco generate the call to C it keep as source A and destination C. The problem is that A is not allowed as outgoing number from the PRI in this way the switch take the default number D as source and the customer on C see the D number as originator. This is not a wanted behaviour becouse the user is not able to know he is calling him. I know that there is a way to change the answer of SIP to insert B as source and C as destination and A as CC-Diversion in the sip request. In this way the cisco will put in the ISDN setup the redirecting number A and the switch box will interpret it as source. I have not so much experience in SIP Proxy to do it. I hope someone can help me. thanks, regards particularly to Ulrich that helped me a lot in the past. Bye, MArcello