I have an issue where Asterisk 1.8 will add or remove the destination port in R-URI, we need this rport to identify the incoming call and route it to correct destination within customer network. I have investigated the working and non working register messages and there is no difference in sip signaling, looks like its asterisk discretion to add or remove port. When an invite is sent to the registerd user we need a rport information in R-uri . the most frustrating bit is that it will work sometimes . I am wondering if this is any known issue or somone knows what needs to be done to fix this issue.