<div dir="ltr"><br><div>Many thanks, I've managed to handle a call properly.</div><div><br></div><div>I really want to thank Daniel, and all that have helped on this subject. </div><div><br></div><div>Regards</div><div>J.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jan 8, 2018 at 11:42 PM, Joel Serrano <span dir="ltr"><<a href="mailto:joel@gogii.net" target="_blank">joel@gogii.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div dir="auto">Just a hint here, try setting $du and then t_relay...</div><div><div class="h5"><br><div class="gmail_quote"><div>On Mon, Jan 8, 2018 at 11:55 Jean Cérien <<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><br><div>Hi</div><div><br></div><div>While I'm trying to get the provider fix the SBC, I am implementing the workaround. </div><div><br></div><div>Almost done here, storing and retrieving the correct address is fine, but when I set my $ru to the corrected value (asterisk IP), the t_relay still sends the packet to the kamailio IP - </div><div><br></div><div>$ru="sip:number@asteriskip:<wbr>5060";<br></div><div>if (!t_relay()) {<br></div><div>....</div><div><br></div><div>Why would the t_relay forward to the kamailio IP and not the asterisk ?</div><div><br></div><div>Rgds</div></div><div><div>J</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jan 8, 2018 at 11:49 AM, Jean Cérien <span><<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12.8px">Many, many thanks !</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">I've posted the full dialog unredacted here: <a href="https://pastebin.com/EE9iwgZf" target="_blank">https://pastebin.com/<wbr>EE9iwgZf</a></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">The OK from Kamailio back to VOIP provider has</div><div style="font-size:12.8px">Record-Route: <sip:KAMAILIOIP;lr=on;ftag=<wbr>SD2rbta01-8dd0e72b-0016-0379-<wbr>0000-0000;did=e0c.e0a1><br></div><div style="font-size:12.8px">Contact: <sip:NUMBER@ASTERISKIP:5060></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">So my understanding is that the ACK should be </div><div style="font-size:12.8px">ACK sip:NUMBER@ASTERISKIP:5060 SIP/2.0</div><div style="font-size:12.8px">....</div><div style="font-size:12.8px">and not</div><div style="font-size:12.8px">ACK sip:NUMBER@KAMAILIOIP:5060 SIP/2.0</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Am I understanding correctly ?</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Rgds</div><span class="m_-3712074151695860991m_5282145479837421489HOEnZb"><font color="#888888"><div style="font-size:12.8px">J</div><div class="m_-3712074151695860991m_5282145479837421489m_1775844981210581009gmail-yj6qo m_-3712074151695860991m_5282145479837421489m_1775844981210581009gmail-ajU" style="margin:2px 0px 0px;font-size:12.8px"></div></font></span></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="m_-3712074151695860991m_5282145479837421489h5">On Mon, Jan 8, 2018 at 11:28 AM, Sebastian Damm <span><<a href="mailto:damm@sipgate.de" target="_blank">damm@sipgate.de</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="m_-3712074151695860991m_5282145479837421489h5"><div>Hi,<br><div class="gmail_extra"><br><div class="gmail_quote"><span>On Mon, Jan 8, 2018 at 2:39 PM, Jean Cérien <span><<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><br><div>Thanks for this answer. The voip provider is not really eager to alter its SBC as it considers that the contact field is not mandatory in the ACK. The RFC states (section 8.1.1.8)</div></div></blockquote><div> </div></span><div>The problem is not that the ACK doesn't carry a Contact header. The problem is that the ACK is constructed incorrectly. This is what the RfC says to UAC behavior (section 12.1.2):</div><div><br></div></div></div><blockquote style="margin:0 0 0 40px;border:none;padding:0px"><div class="gmail_extra"><div class="gmail_quote"><div>The route set MUST be set to the list of URIs in the Record-Route
header field from the response, taken in reverse order and preserving
all URI parameters. If no Record-Route header field is present in
the response, the route set MUST be set to the empty set. This route
set, even if empty, overrides any pre-existing route set for future
requests in this dialog. <b>The remote target MUST be set to the URI
from the Contact header field of the response.</b> </div><div><br></div></div></div></blockquote>This is what the carrier's SBC gets wrong. It doesn't address your Asterisk but instead addresses your Kamailio, although the Contact of your 200 OK (hopefully) contains the Asterisk IP.<div class="gmail_extra"><div class="m_-3712074151695860991m_5282145479837421489m_1775844981210581009m_2516756081510492400gmail_signature"></div>
</div><div><br></div><div>Please verify that your 200 OK going to the carrier actually does carry a Contact header with the Asterisk IP, but if it does, section 12.1.2 of the SIP RfC could help when arguing with the carrier.</div><div><br></div><div>Regards</div><span class="m_-3712074151695860991m_5282145479837421489m_1775844981210581009HOEnZb"><font color="#888888"><div>Sebastian</div></font></span></div>
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