[Serusers] Asterisks to ser to asterisk (voicemail)
Dave Bath
dave at fuuz.com
Wed Jul 28 22:39:48 CEST 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 at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
Behalf Of GR S
Sent: 28 July 2004 21:31
To: jon at bostontech.com
Cc: serusers at lists.iptel.org
Subject: Re: [Serusers] Asterisks to ser to asterisk (voicemail)
Hello,
--- jon at 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 at yahoo.com>
> 07/28/2004 04:02 PM
>
> To: jon at bostontech.com, serusers at lists.iptel.org
> cc: oej at edvina.net, andres at telesip.net
> Fax to:
> Subject: Re: [Serusers] Asterisks to ser to asterisk
> (voicemail)
>
>
> Hello,
>
> --- "Olle E. Johansson" <oej at 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 at yahoo.com>
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