<div dir="ltr">Regards for everybody<br><br>Joseph your solution is very interesting, but I think I will have another two problems the first trouble is with billing, Because if I answer the call it will be charged right?<br>
And the second trouble is when I send the Hangup the asterisk will drop the call right?<br><br><br><br><br><div class="gmail_quote">2008/9/2 Joseph <span dir="ltr"><<a href="mailto:tech@ekn.com">tech@ekn.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d">On 09/02/08, Rony Ron wrote:<br>
> Hi,<br>
> imho you can do it with call forward,<br>
> you receive the number<br>
> you check the database if the number is there<br>
> then forward to the new number (prefixing it with what ever you want)<br>
> BR,<br>
<br>
</div>There is a way to redirect your call back to the central(Ericsson AXE)<br>
instead of keeping the media in your path.<br>
<br>
Here is a sample:<br>
<br>
exten => _X.,1,Set(_SS7_LSPI_IDENT=ON)<br>
exten => _X.,n,Set(_SS7_RLT_ON=YES)<br>
exten => _X.,n,Answer()<br>
exten => _X.,n,Playback(demo-congrats)<br>
<br>
<Do your database lookup here and than redirect the call<br>
back to the ss7 switch based on your lookup results<br>
and drop out of the media path><br>
<br>
exten => _X.,n,Dial(ZAP/r2/8005551212,30)<br>
exten => _X.,n,Hangup()<br>
<div class="Ih2E3d"><br>
<br>
<br>
><br>
> Virmones Pereira a écrit :<br>
> > Hi,<br>
> ><br>
> > I would like to use asterisk with SS7 as a STP for Number Portability<br>
> > GW, the idea of the system is follow:<br>
> ><br>
> > When the SS7 central(Ericsson AXE) receive the call this should be<br>
> > route to the Asterisk to trigger the number portability database by<br>
> > SS7/ISUP method if the asterisk found this destination number in the<br>
> > number portability database asterisk will insert the Routing Number in<br>
> > the begin of the called number and then route back this call to the<br>
> > SS7 central.<br>
> ><br>
> > Ex:<br>
> ><br>
> > user dial 551132323232 this call go the asterisk and asterisk turn<br>
> > back with 55112551132323232.<br>
> ><br>
> > I wanna do this operation using asterisk as a STP where the SS7 use<br>
> > only the signaling channel, the media should go directly to the SSP<br>
> ><br>
> > somebody knows how to do it?<br>
> ><br>
> ><br>
<br>
</div>--<br>
respectfully,<br>
<font color="#888888">Joseph<br>
</font><br>-----BEGIN PGP SIGNATURE-----<br>
Version: GnuPG v1.4.9 (GNU/Linux)<br>
<br>
iEYEARECAAYFAki9GN4ACgkQ5CyZqOno04xQ5ACfdnzkePZP4Ubt/A20ZuK6o9E9<br>
iS0AniIN8HWmn3iASu7VU3RA8zNqJQOl<br>
=rRpr<br>
-----END PGP SIGNATURE-----<br>
<br>_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-ss7 mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br></blockquote></div><br></div>