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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