26 Sep
2004
26 Sep
'04
10:37 a.m.
On Sep 25, 2004 at 18:52, Java Rockx <javarockx(a)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(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?
I think you want to keep the result of lookup("aliases"), since in this
case the original is 4075551234(a)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