<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><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>
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