<div>Hi, </div>
<div>I managed to install my patton gateway but i did not succeed to pass the caller id to the sip phone on incoming calls ...</div>
<div>instead i see call from 105 ( which is the sip client extension of the patton ) </div>
<div> </div>
<div>do you know the way to pass the caller id of the caller to the ip phone iso the gw extension on incoming calls ?</div>
<div>thanks !</div>
<div>jean-louis<br><br> </div>
<div><span class="gmail_quote">2006/11/4, Guido Hecken <<a href="mailto:guido.hecken@gwsnettech.de">guido.hecken@gwsnettech.de</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi Kevin,<br><br>you have to create a gateway in the Smart Node:<br><br>gateway sip sip<br>bind interface eth1 router
<br><br>service default<br> domain gwsnettech.local<br> realm gwsnettech.local<br> authentication isdngw2 password huffvtzddzdjkhuztztufuz== encrypted<br>default<br> default-server hallinux2.gwsnettech.local 5060 loose-router
<br> registrar hallinux2.gwsnettech.local 5060<br><br>In sip.conf, something like this:<br><br>[isdngw2]<br>type=friend<br>username=isdngw2<br>secret=the_unencrypted_password_from _above<br>host=<a href="http://192.168.161.135">
192.168.161.135</a><br>;host=dynamic<br>fromuser=gwsnettech<br>fromdomain=gwsnettech.local<br>nat=no<br>context=isdngw2-in<br>canreinvite=no<br><br>If you need a complete running config, I can send it to you offlist.<br><br>
Hope, it helps...<br><br><br>Guido<br><br>________________________________________<br>Von: Kevin Withnall [mailto:<a href="mailto:kevin@ilb.com.au">kevin@ilb.com.au</a>]<br>Gesendet: Samstag, 4. November 2006 05:58<br>An:
<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>Betreff: [asterisk-users] Patton 1400<br><br>I have a patton 1400 setup to handle the bri interface. As a trixbox user, I<br>wanted a sip trunk rather than having to re-compile bri support into
<br>trixbix.<br><br>Anyway, I have it working now so that asterisk can make calls and they are<br>passed properly to the telephone network. Incoming calls however are another<br>matter. I have (after turning on cli debug in the 1400) determined that its
<br>getting stuck in the routing system. I don't know what destination to make<br>to get it to sip connect to the asterisk box.<br><br>Ive tried making an interface that has a remote address of the asterisk box<br>but that doesn't work. Can someone send me a config for anything they have
<br>done that is similar to this ? I believe the Patton 1200 is also the same<br>unit apart from extra ports.<br><br>Any help would be greatly appreciated.<br><br>Regards<br>Kevin<br><br>_______________________________________________
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