<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>And stupid n00b me – left the IP addresses in this email… crash and burn…</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> Robert <<a href="mailto:robert@inteli-core.com">robert@inteli-core.com</a>><br><span style="font-weight:bold">Reply-To: </span> OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br><span style="font-weight:bold">Date: </span> Wed, 07 Mar 2012 14:55:18 -0500<br><span style="font-weight:bold">To: </span> <<a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br><span style="font-weight:bold">Subject: </span> [OpenSIPS-Users] Simple Forwarding Proxy<br></div><div><br></div><div><div 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>I am trying to install OpenSIPS as a simple SIP proxy to allow users to register on an offset port – which then gets re-directed to our VoIP platform.</div><div><br></div><div>As far as registrations go – I have that working.</div><div><br></div><div>I have added a binding for port 53 and port 8080</div><div>I also added rewritehostport statements in the config as per guides for Asterisk integration.</div><div><br></div><div>And I can complete calls outbound and get 2 way audio.</div><div>The issue I am experiencing is that I cannot get inbound calls to come through.</div><div><br></div><div>I'm not sure if this a NAT issue with the UA – or if possibly I have not added enough headers in the config to allow the VoIP platform to know that the UA is behind NAT.</div><div><br></div><div>I'm a n00b at OpenSIPS – so please bare with me :)</div><div><br></div><div><div> # account only INVITEs</div><div> if (is_method("INVITE")) {</div><div> setflag(1); # do accounting</div><div> rewritehostport("xxx.xxx.xxx.xxx:5060");</div><div> }</div></div><div><br></div><div><div> if (is_method("REGISTER"))</div><div> {</div><div> # authenticate the REGISTER requests (uncomment to enable auth)</div><div> ##if (!www_authorize("", "subscriber"))</div><div> ##{</div><div> ## www_challenge("", "0");</div><div> ## exit;</div><div> ##}</div><div> ##</div><div> ##if (!db_check_to())</div><div> ##{</div><div> ## sl_send_reply("403","Forbidden auth ID");</div><div> ## exit;</div><div> ##}</div><div> save("location");</div><div> rewritehostport("xxx.xxx.xxx.xxx:5060");</div><div> route(1);</div><div><br></div><div> if (!save("location"))</div><div> sl_reply_error();</div><div><br></div><div> exit;</div><div> }</div></div><div><br></div><div><div>route[1] {</div><div> # for INVITEs enable some additional helper routes</div><div> if (is_method("INVITE")) {</div><div> rewritehostport("xxx.xxx.xxx.xxx:5060");</div><div> t_on_branch("2");</div><div> t_on_reply("2");</div><div> t_on_failure("1");</div><div> }</div><div><br></div><div> if (!t_relay()) {</div><div> sl_reply_error();</div><div> };</div><div> exit;</div><div>}</div></div><div><br></div></div></div>
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