<br><font size=2 face="sans-serif">it looks like i will have to do that,
but it's a little sloppy. oh well, that's life.</font>
<br>
<br><font size=2 face="sans-serif">-Jon</font>
<br>
<br>
<br>
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<td><font size=1 face="sans-serif"><b>GR S <gr_sh2003@yahoo.com></b></font>
<p><font size=1 face="sans-serif">07/28/2004 04:30 PM</font>
<td><font size=1 face="Arial"> </font>
<br><font size=1 face="sans-serif"> To:
jon@bostontech.com</font>
<br><font size=1 face="sans-serif"> cc:
serusers@lists.iptel.org</font>
<br><font size=1 face="sans-serif"> Fax
to: </font>
<br><font size=1 face="sans-serif"> Subject:
Re: [Serusers] Asterisks to ser to asterisk
(voicemail)</font></table>
<br>
<br>
<br><font size=2><tt>Hello,<br>
<br>
--- jon@bostontech.com wrote:<br>
<br>
> yes, i know that this will work, but the issue is that not every sip
user <br>
> who is called has voicemail. I want SER to determine who should be
<br>
> rerouted or who shouldn't.<br>
<br>
Still you dont need to worry. Let all un-attended calls come back to Asterisk.
It will drop the<br>
calls if it can't find a mail box number. May not be the right method,
though.<br>
<br>
> -Jon<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> GR S <gr_sh2003@yahoo.com><br>
> 07/28/2004 04:02 PM<br>
> <br>
> To: jon@bostontech.com,
serusers@lists.iptel.org<br>
> cc: oej@edvina.net, andres@telesip.net<br>
> Fax to: <br>
> Subject: Re:
[Serusers] Asterisks to ser to asterisk <br>
> (voicemail)<br>
> <br>
> <br>
> Hello,<br>
> <br>
> --- "Olle E. Johansson" <oej@edvina.net> wrote:<br>
> <br>
> > Andres wrote:<br>
> > <br>
> > > <br>
> > >><br>
> > >> My question is, is there any way to have ser receive
a call from <br>
> > >> asterisk and then reroute it back to the same asterisk
server without <br>
> <br>
> > >> getting a "loop detected" error?<br>
> > >><br>
> > > Aren't you seeing this "loop detected" on the
Asterisk CLI?? If so <br>
> > > should post this in the Asterisk list instead. We
know this happens <br>
> > > anytime you try to loop a call back to Asterisk, but its
Asterisk who <br>
> > > complains. Not SER.<br>
> > > <br>
> > Answer from the Asterisk users list :-)<br>
> > <br>
> > No, there's not a way to do it, but maybe to issue a 302 redirect.<br>
> > Haven't tried it, but that may work.<br>
> > <br>
> > The Loop Detected stuff is annoying, yes.<br>
> > <br>
> > /O<br>
> > <br>
> <br>
> >From a great fan of Asterisk and SER :-)<br>
> <br>
> I am not sure about the exact problem, but there is another way to
acheive <br>
> this. You dont need to<br>
> re-route the calls back from SER to Asterisk. Set a timeout in the
<br>
> Asterisk Dial statement and<br>
> forward the call to SER. If the callee attends the call, you can talk,
and <br>
> if not, make Asterisk<br>
> forward the call to voicemail when it hits the timeout. I have this
<br>
> feature enabled in a local<br>
> system running SER on 5060 and Asterisk on 5070.<br>
> <br>
> Best Regards,<br>
> <br>
<br>
<br>
=====<br>
Girish Gopinath <gr_sh2003@yahoo.com><br>
<br>
<br>
<br>
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