New subject: [Serusers] Asterisks to ser to asterisk (voicemail)

28 Jul 2004


      As you guys seem to have some experience with this area perhaps I will
re-ask my question this thread..
I have SER-->asterisk forwarding working fine now, and I have divert on
unavailable working.  However, I have two outstanding problems:
(1) If a user is called with their alphanumeric ID instead of their
numerical alias, * does not pick up the call.  This is as expected, as
the dial pattern in * is _[1-9] [0-9] [0-9] [0-9].  However, it must be
fairly common to call people with their email addresses for example...
so how can I make ser pass the alias to * instead of the alpha URI?
(2) If a user is offline, I get a 404 immediately, instead of anything
else - for example diverting immediately to vm.  I don't quite
understand this at the moment.. as I have the t_on_failure set up before
the location lookups... does the t_on_failure not catch 404 failures?
Any thoughts would be very much appreciated.... anything I can provide,
please let me know...
Thanks again everyone,
Dave
-----Original Message-----
From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of GR S
Sent: 28 July 2004 21:31
To: jon@bostontech.com
Cc: serusers@lists.iptel.org
Subject: Re: [Serusers] Asterisks to ser to asterisk (voicemail)
Hello,
--- jon@bostontech.com wrote:
...
yes, i know that this will work, but the issue is that not every sip
user
...
who is called has voicemail. I want SER to determine who should be 
rerouted or who shouldn't.
Still you dont need to worry. Let all un-attended calls come back to
Asterisk. It will drop the
calls if it can't find a mail box number. May not be the right method,
though.
...
-Jon
GR S gr_sh2003@yahoo.com
07/28/2004 04:02 PM
    To:     jon@bostontech.com, serusers@lists.iptel.org
    cc:     oej@edvina.net, andres@telesip.net
    Fax to: 
    Subject:        Re: [Serusers] Asterisks to ser to asterisk 

(voicemail)
Hello,
--- "Olle E. Johansson" oej@edvina.net wrote:
...
Andres wrote:
...
...
My question is, is there any way to have ser receive a call from 
asterisk and then reroute it back to the same asterisk server
without
...
...
...
...
getting a "loop detected" error?
Aren't you seeing this "loop detected" on the Asterisk CLI??  If
so
...
...
...
should post this in the Asterisk list instead.  We know this
happens
...
...
...
anytime you try to loop a call back to Asterisk, but its Asterisk
who
...
...
...
complains.  Not SER.
Answer from the Asterisk users list :-)
No, there's not a way to do it, but maybe to issue a 302 redirect.
Haven't tried it, but that may work.
The Loop Detected stuff is annoying, yes.
/O
...
From a great fan of Asterisk and SER :-)
I am not sure about the exact problem, but there is another way to
acheive
...
this. You dont need to
re-route the calls back from SER to Asterisk. Set a timeout in the 
Asterisk Dial statement and
forward the call to SER. If the callee attends the call, you can talk,
and
...
if not, make Asterisk
forward the call to voicemail when it hits the timeout. I have this 
feature enabled in a local
system running SER on 5060 and Asterisk on 5070.
Best Regards,
=====
Girish Gopinath  gr_sh2003@yahoo.com
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RE: [Serusers] Asterisks to ser to asterisk (voicemail)