[Serusers] SER + Asterisk Voicemail -- Seeking Advise

[Andrei Pelinescu-Onciul]

On Sep 25, 2004 at 18:52, Java Rockx <javarockx at 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 at mycompany.com with a PSTN alias 4075551234. In addition this user has an
> Asterisk mailbox configured as 1000 at mycompany.com
> 
> When someone dials sip:1000 at 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 at 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 at 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 at mycompany.com and you get
sip:1000 at 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