On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > Using pjsip 2.7.2 on Asterisk 15.5 > Really struggling to make sense of translating these old 1.8 SIP > instructions into a neat pjsip_wizard conf suitable for 2018 > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > In pjsip_wizard.conf, I have the following, which seems to get me > registered, and it responds to an incoming call, but I always get > this: > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > log_failed_request: Request 'INVITE' from '"demo" > <sip:myusern...@sip2sip.info>' failed for 'x.x.x.x:5060' (callid: > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > here's what I have in pjsip_wizard.conf > > [sip2sip] > type = wizard > sends_auth = yes > accepts_registrations = yes > transport = simpletrans > outound_auth/username = myusern...@sip2sip.info > outound_auth/password = password > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > endpoint/allow = alaw > endpoint/context = fromsip2sip > aor/max_contacts = 3 > registration/contact_user = myusername > outbound_proxy = proxy.sipthor.net > endpoint/language=en_GB
This is an ITSP trunk, you've configured it kind of as if it were a phone. Instead of "accepts_registrations" you likely want "sends_registrations". Asterisk needs to register to them. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users