<div>Thank you!, I will try that in a few hours and let you know what happens.</div>
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<div class="gmail_quote">On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini <span dir="ltr"><<a href="mailto:marcotasto@libero.it">marcotasto@libero.it</a>></span> wrote:<br>
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<div class="Wj3C7c"><br><br>Pascal Bruno wrote:
<blockquote cite="http://midb09af520901160658m2cb1970cr397b1fd996a85f44@mail.gmail.com" type="cite">
<div>Thanks for your reply!</div>
<div> </div>
<div>Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register.</div>
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<div class="gmail_quote">On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini <span dir="ltr"><<a href="mailto:marcotasto@libero.it" target="_blank">marcotasto@libero.it</a>></span> wrote:<br>
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<div bgcolor="#ffffff" text="#000000">Emmanuel Pascal Bruno wrote:
<blockquote cite="http://midb09af520901151700m4b9117d2sd9b5aa74870924f5@mail.gmail.com" type="cite">
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<div>Has anyone been able to configure portech's mv-378 gateway with asterisk?</div>
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<div>I did the configuration as per the manual but it does not work.</div>
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<div>My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register.</div>
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<div>The gateway and the asterisk box are on two different location (two network, 2 differrent IP address).</div>
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<div>I would appreciate any kind of tutorial or advice on how to make it work.</div>
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<div>Thanks</div></div></div><pre><hr width="90%" size="4">
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<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre></blockquote><br>Hi, <br>I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file.<br>
<br>[GSMGtw1]<br>type=friend<br>context=from-gsm<br>host=dynamic ; we have a DHCP assigned address<br>secret=reallyverysecret<br>nat=no ; there is not NAT between phone and Asterisk<br>
canreinvite=no <br>dtmfmode=INFO<br>insecure=invite ; required to overcome authentication problems in incoming calls<br>call-limit=1 ; permit only 1 outgoing call at a time<br>disallow=all<br>
allow=ulaw <br>allow=alaw <br>allow=gsm<br>qualify=500<br><br>I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech.<br>
<br>Best regards,<br>Marco Signorini.<br><br></div><br></blockquote></div></blockquote><br></div></div>Hi,<br><br>I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. <br>
For this reason:<br>1. The only rule I've on Mobile to Lan is CID=*; <a href="mailto:URL=mob@192.168.0.5" target="_blank">URL=mob@192.168.0.5</a> where "mob" is the extension I've generated in the asterisk box under the context where the Portech operates;<br>
2. The only rule I've on Lan to Mobile is URL=*; Call Num=#<br><br>I think the most relevant parameters for your problem are under the "Service Domain" menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that:<br>
<br>UserName: GSMGtw1<br>RegisterName: GSMGtw1<br>RegisterPassword: reallyverysecret<br>Domain Server: 192.168.0.5<br>Proxy Server: 192.168.0.5<br><br>Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set:<br>
<br>domain=192.168.0.5<br><br>on sip.conf [general] section (remember to issue a sip reload from asterisk cli).<br><br>Hope this helps!<br><br><br>Best regards.<br>Marco Signorini<br><br><br><br>========================<br>
Marco Signorini<br>INGEGNI Tech S.r.l.<br><a href="http://www.ingegnitech.com/" target="_blank">http://www.ingegnitech.com</a><br></div><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
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