<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">oh.. its simple.<div><br></div><div>"[res_pjsip_endpoint_identifier_ip]" should be before "identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]”</div><div><br></div><div>Thanks all for help :)</div><div><br></div><div><div><div>On 17 Jul 2014, at 11:05, Nick Awesome <<a href="mailto:jleed@me.com">jleed@me.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><meta http-equiv="Content-Type" content="text/html charset=windows-1252"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">New information, as I said I’m using realtime,<div>thats the problem!</div><div><br></div><div>I just tested using static config file and it is working perfect.</div><div>after some research I figured out that problem with “ps_endpoint_id_ips" for some reason asterisk ignoring matches in this table,</div><div><br></div><div>I have string in sorcery.conf</div><div><br></div><div>identify = realtime,ps_endpoint_id_ips</div><div><br></div><div><div>also have string in extconfig.conf</div><div><br></div><div>ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips</div><div><br></div><div>and ofc I have table</div><div><br></div><div><div>CREATE TABLE `pbx_endpoint_id_ips` (</div><div> `id` varchar(40) NOT NULL,</div><div> `endpoint` varchar(40) DEFAULT NULL,</div><div> `match` varchar(80) DEFAULT NULL,</div><div> UNIQUE KEY `id` (`id`),</div><div> KEY `ps_endpoint_id_ips_id` (`id`)</div><div>) ENGINE=InnoDB DEFAULT CHARSET=latin1;</div></div><div><br></div><div>with entry </div><div><br></div><div>10001 | 10001 | 85.195.98.178</div><div><br></div><div>but thats just didn’t works(</div><div><br></div><div>is this a bug and should I open ticket ?</div><div><br></div><div><div>On 16 Jul 2014, at 21:13, Nick Awesome <<a href="mailto:jleed@me.com">jleed@me.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><meta http-equiv="Content-Type" content="text/html charset=us-ascii"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Ok there is my test account from <a href="http://sipiko.net/">sipiko.net</a><div><br></div><div>username: cb5069</div><div>password: sqv664yqtp</div><div>domain: <a href="http://callme.sipiko.net/">callme.sipiko.net</a></div><div><br></div><div>its using username/password authentication.</div><div>because its just website widget I need only inbound calls from this peer,</div><div>test call can be done from url: </div><div><a href="http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes">http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes</a></div><div><br></div><div>on my side I have an asterisk 12 using pjsip</div><div><br></div><div>Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context</div><div><br></div><div>help if you can please:)</div><div><br></div><div><div><div>On Jul 16, 2014, at 8:53 PM, Joshua Colp <<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Nick Awesome wrote:<br><blockquote type="cite">I thought that<br><blockquote type="cite"><blockquote type="cite">type=identify<br></blockquote></blockquote>will match an IP address and accept it,<br><br>well, in my example I can control both sides and able to configure it<br>without registration. in real life I have a provider that requires<br>username/password authentication<br><br>provider gives me - Username - Password - DomainName<br></blockquote><br>They may require it for *outgoing* calls to them but for incoming I<br>highly doubt they'd want you to authenticate them. It's usually always<br>IP authentication.<br><br><blockquote type="cite">I have configure it like I showed before and have exactly the same<br>notice<br><br>[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246<br>log_unidentified_request: Request from<br>'"cb5069"<<a href="sip:asterisk@85.195.98.178">sip:asterisk@85.195.98.178</a>>' failed for<br>'85.195.98.178:5060' (callid:<br><a href="mailto:173995aa2e25283807700d65055c9214@85.195.98.178">173995aa2e25283807700d65055c9214@85.195.98.178</a>) - No matching<br>endpoint found 85.195.98.178 is an operator,<br><br>so what I should add to my config to be able accept calls from<br>Registered peer ?<br></blockquote><br>The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.<br><br>Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.<br><br>-- <br>Joshua Colp<br>Digium, Inc. | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>Check us out at: <a href="http://www.digium.com/">www.digium.com</a> & <a href="http://www.asterisk.org/">www.asterisk.org</a><br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div></div>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div></div>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div></body></html>