Ok there is my test account from sipiko.net username: cb5069 password: sqv664yqtp domain: callme.sipiko.net
its using username/password authentication. because its just website widget I need only inbound calls from this peer, test call can be done from url: http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes on my side I have an asterisk 12 using pjsip Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context help if you can please:) On Jul 16, 2014, at 8:53 PM, Joshua Colp <[email protected]> wrote: > Nick Awesome wrote: >> I thought that >>>> type=identify >> will match an IP address and accept it, >> >> well, in my example I can control both sides and able to configure it >> without registration. in real life I have a provider that requires >> username/password authentication >> >> provider gives me - Username - Password - DomainName > > They may require it for *outgoing* calls to them but for incoming I > highly doubt they'd want you to authenticate them. It's usually always > IP authentication. > >> I have configure it like I showed before and have exactly the same >> notice >> >> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 >> log_unidentified_request: Request from >> '"cb5069"<sip:[email protected]>' failed for >> '85.195.98.178:5060' (callid: >> [email protected]) - No matching >> endpoint found 85.195.98.178 is an operator, >> >> so what I should add to my config to be able accept calls from >> Registered peer ? > > 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. > > Without further details of your setup (IP addresses, who are calling who) and > how you want it to work I can't answer. > > -- > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
