On Sep 25, 2004 at 18:52, Java Rockx javarockx@yahoo.com wrote:
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@mycompany.com with a PSTN alias 4075551234. In addition this user has an Asterisk mailbox configured as 1000@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@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?
I think you want to keep the result of lookup("aliases"), since in this case the original is 4075551234@mycompany.com and you get sip:1000@mycompany.com only after lookup("aliases"). So it should look somehting like: lookup("aliases"); if (!lookup("location")){ # try voicemail, user not registered break; }
and in your failure route: revert_uri(); lookup("aliases"); # forward to voicemail, user not responding/busy
Andrei