Hello All. I finally have my lookup("aliases") working thanks to Zeus Ng. Now that users can have aliases on my ser proxy I have a question regarding voicemail. I'm hoping someone can give me an idea of how best to address this issue. I use ser for all SIP stuff and Asterisk for voicemail only. I have ser and asterisk working nicely together. A typical scenerio would be like this. I have a ser user named 1000(a)mycompany.com with a PSTN alias 4075551234. In addition this user has an Asterisk mailbox configured as 1000(a)mycompany.com When someone dials sip:1000@mycompany.com and there is no answer they will get sent to voicemail, which then Asterisk will say "The user at extension one-zero-zero-zero is unavailable. Please leave your message after the tone..." But what happens when a caller dials 4075551234(a)mycompany.com and gets routed to voicemail? 4075551234 doesn't exist in asterisk. If I use lookup("aliases") in my ser.cfg routing plan can I revert back to the original sip:1000@mycompany.com before sending the caller off to the asterisk voicemail? If so, how? Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail