<html><head></head><body bgcolor="#FFFFFF"><div>Kamailio's default conf is really a good place to start, you just have to disable the authentication, by commenting the corresponding route.<br><br>You can start by looking at a wireshark/tcpdump dump on your PortaOne server to know where the rtp stream is going, and then troubleshoot accordingly..</div>
<div><br></div><div>On your PortaOne server, what is the location of registered users? Is it their IP@, or that of OpenSips server?<br><div><br></div>Reda</div><div><br>On 9 mars 2012, at 16:28, Robert <<a href="mailto:robert@inteli-core.com">robert@inteli-core.com</a>> wrote:<br>
<br></div><div></div><blockquote type="cite"><div><div>Thank you so much for answering.. The folks at the other place don't want to help.</div><div><br></div><div>We have a PortaOne SIP platform… It does everything… It's listening on port 5060 – public ip..</div>
<div><br></div><div>We want to deploy Kamailio or some derivative of SER to listen on PUBLIC port – other than 5060 – for our customers who cannot reach 5060.</div><div><br></div><div>Using a very simple rewritehostport statement – I was successful in getting the registration to pass through. I.e. Registered my Aastra and X-Lite to port 8080 and saw registration in PortaOne.</div>
<div><br></div><div>I do not want K/SER to do actual auth / registration – I.e. I do not care who tries to use it – I want final registration to be PortaOne… That's why I think the simple rewritehostport worked well. OpenSIPS was not doing any auth – just simply passing it on.</div>
<div><br></div><div>Our PortaOne was handling RTP Proxy – so during an outbound call I was getting 2 way audio – no problem.</div><div>However when an inbound call would come in from PortaOne – it was being delivered to OpenSIPS and looks like Opensips was reporting 401 Unauthorized.</div>
<div><br></div><div>The Aastra / X-Lite are NAT… It's possible that either I need to look at nathlper closer – or that I misconfigured something somewhere.</div><div><br></div><div>I guess I'm not fluent enough in the configs and options to get a good config.</div>
<div><br></div><div><br></div><span id="OLK_SRC_BODY_SECTION"><div style="font-family:Calibri;font-size:11pt;text-align:left;color:black;BORDER-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADDING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:medium none;PADDING-TOP:3pt">
<span style="font-weight:bold">From: </span> Reda Aouad <<a href="mailto:reda.aouad@gmail.com">reda.aouad@gmail.com</a>><br><span style="font-weight:bold">Reply-To: </span> "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br>
<span style="font-weight:bold">Date: </span> Fri, 9 Mar 2012 15:45:14 +0100<br><span style="font-weight:bold">To: </span> "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br>
<span style="font-weight:bold">Subject: </span> Re: [SR-Users] Proxy Registrations offset port<br></div><div><br></div><div dir="ltr"><font color="#3366ff"><font><font face="tahoma,sans-serif">Hi Robert,</font></font></font><div>
<font color="#3366ff"><font><font face="tahoma,sans-serif"><br></font></font></font></div><div><font color="#3366ff"><font><font face="tahoma,sans-serif">I had the same feeling about O***sips community.....</font></font></font></div>
<div><font color="#3366ff"><font><font face="tahoma,sans-serif"><br></font></font></font></div><div><font color="#3366ff"><font><font face="tahoma,sans-serif">I have a similar setup. A proxy that listens on port other than 5060, forwarding to other servers on 5060.</font></font></font></div>
<div><font color="#3366ff"><font><font face="tahoma,sans-serif">If you can better define better your problem with your setup, maybe we can help you better.</font></font></font></div><div><font color="#3366ff"><font><font face="tahoma,sans-serif"><br>
</font></font></font></div><div><font color="#3366ff"><font><font face="tahoma,sans-serif">Is your server listening only on one port 8080? Is your server behind NAT or does it have a public IP @ on the interface it's listening on? What is the exact problem with inbound calls? Are you doing record-route? Do you forward only registrations, or are your handling invites as well? What server listens on port 5060? Another Kamailio server? Does it handle clients behind NAT?</font></font></font></div>
<div><font color="#3366ff"><font><font face="tahoma,sans-serif"><br></font></font></font></div><div><font color="#3366ff"><font><font face="tahoma,sans-serif">You should know that Kamailio is a SIP server, not a PBX/sofswitch/SBC as Asterisk or Freeswitch.</font></font></font></div>
<div><font color="#3366ff" face="tahoma,sans-serif">If you define better your problem, and send me your config file (by private email if you like), I may look at it.</font></div><div><font color="#3366ff" face="tahoma,sans-serif"><br>
</font></div><div><div dir="ltr"><div><font color="#3366ff" face="tahoma,sans-serif">Reda</font></div></div><br><br><br><div class="gmail_quote">On Fri, Mar 9, 2012 at 15:32, Robert <span dir="ltr"><<a href="mailto:robert@inteli-core.com">robert@inteli-core.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="font-size:14px;font-family:Calibri,sans-serif;word-wrap:break-word"><div>I've been playing with OpenSIPS and am willing to change over if it means I'll get a more informed answer from this community..</div>
<div><br></div><div>I need to set up a proxy forwarding server that can listen on a port other than 5060 – I.e. 8080 and forward the registration requests to another Voip platform we have that is running on 5060.</div><div>
<br></div><div>I've been able to this under OpenSIPS but I'm having issues with inbound calls. It could be a NAT issue.</div><div><br></div><div>Basically listen on 8080 – forward any and all registrations to x:5060 — which I've done using rewritehostport</div>
<div><br></div><div>Anyone have any good example configs for this? I'm a n00b at this OpenSER stuff – but have been around asterisk / freeswitch / portaone for a while.</div><div><br></div><div>Thanks in advance</div>
</div><br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br><a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br></blockquote></div><br></div></div>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</span>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list</span><br><span><a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a></span><br>
<span><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a></span><br></div></blockquote></body></html>