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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Cumhur ,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thank you for your reply ,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>IP to IP not supported OS for As5400 ( XM supported ) also
I have tried on Soft-switchs ( MERA / VMSM / ) applying user information from
the anonymous ingress calls than translations.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Soft-switchs binding their Localhost.localdomain information to
at the end of the packet its shown like this at the from side of contact. I
found your sended documentation on the forum I read than I posted my details.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Invite packet<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> From
: uri <a href="sip:callingline@calleripaddress">sip:callingline@calleripaddress</a><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>
Contach uri : user@calleripaddress -- > it should be
Username@calledDomain or outbound proxy actually its not irregular format all soft
phones working like this if you need to send user name for requested uri
and also if you are directly sending user name to target proxy it s automatically
binding Proxy ip address as a calledDomain<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>May be you can suggest any idea after those explanations<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Bilay<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Cumhur Kizilari
[mailto:cumhur@gmail.com] <br>
<b>Sent:</b> Sunday, March 18, 2007 12:45 PM<br>
<b>To:</b> bilay ozer; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP2SIP2PROXY<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>Hi,<br>
<span style='color:#1F497D'>s </span><br>
Are you using IP to IP IOS on your router ? if not you should (in my opinion)
use IP to IP IOS for call forwarding over IP.<br>
<br>
<a
href="http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_data_sheet09186a00801da698.html">http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_data_sheet09186a00801da698.html</a><br>
<br>
<a
href="http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_guide_chapter09186a00806b0a14.html">http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_guide_chapter09186a00806b0a14.html</a><br>
<br>
Regards,<br>
Cumhur<br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal><span class=gmailquote>On 3/17/07, <b>bilay ozer</b> <<a
href="mailto:bilayo@a-c-tgmbh.de">bilayo@a-c-tgmbh.de</a>> wrote:</span><o:p></o:p></p>
<p class=MsoNormal>Hi There,<br>
<br>
<br>
I am trying to apply my scenario as a such ways. I tried too many options<br>
with SS but there is no capability for that<br>
<br>
My customer has a SIP-gw they will originate their calls to directly MY<br>
voice GW . My voice-gw will receive this call than will forward to SIP<br>
proxy.<br>
<br>
Sip-as5400-sip with uri outboundproxy<br>
<br>
Here is the conditions.<br>
<br>
Proxy accepting calls with Requested URI mean ( account information user @ <br>
domainofproxy) I have 10 sip account on my hand is there any way to set that<br>
account under the voip dial-peers with destination SIP proxy also accepting<br>
calls with anonymous @ calleripaddress means with translating with source <br>
<br>
I need a configuration example for that task there are no PSTN or trunk<br>
connection on As5400.<br>
<br>
Regards<br>
<br>
<br>
<br>
<br>
<br>
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<p class=MsoNormal><o:p> </o:p></p>
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