It didn't help, I've tested it before...<br><br><div><span class="gmail_quote">2007/11/11, Yann JOUANIN <<a href="mailto:asterisk@intelunix.fr">asterisk@intelunix.fr</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB">Perhaps you could try to
change the Agent name in SIP.conf (replace asterisk by another one, like
Xlite).</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB"> </span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB">Be sure that you asterisk
server has the same configuration than your Xlite client , look at "default expirey"
as an example. My SIP provider need some custom parameters to accept calls.</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB"> </span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB">Regards,</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB"> </span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB">yann</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;" lang="EN-GB"> </span></font></p>
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<p><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">De :</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight: bold;">De la part de
</span></b> Giedrius Augys<br>
<b><span style="font-weight: bold;">Envoyé :</span></b> dimanche 11 novembre
2007 11:39<br>
<b><span style="font-weight: bold;">À :</span></b>
<a href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users@lists.digium.com</a><br>
<b><span style="font-weight: bold;">Objet :</span></b> [asterisk-users]
detect asterisk pbx via sip</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> </span></font></p>
<p><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Hello,<br>
My situation is that , I can't make calls with asterisk, but with x-lite
works fine. Asterisk shows , that successfully registers with another SIP
server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408
Request timeout. And as I said , with x-lite no problems. I heard that for
comercial purposes, this SIP server detects asterisk , and ignores him. Or
maybe it check is it server or device? <br>
Maybe somebody can give me some advices...<br>
Thanks</span></font></p>
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