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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Perhaps you could try to
change the Agent name in SIP.conf (replace asterisk by another one, like
Xlite).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>yann<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>De :</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <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>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Objet :</span></b> [asterisk-users]
detect asterisk pbx via sip</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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<o:p></o:p></span></font></p>
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