When doing an inter-Asterisk call transfer using SIP, I am using the "fromuser" parameter to route the call into the proper context on the receiving server. This causes the original callerid to be lost. Does anyone have any ideas how to preserve the original callerid in this scenario? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.