oh.. its simple. "[res_pjsip_endpoint_identifier_ip]" should be before "identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]”
Thanks all for help :) On 17 Jul 2014, at 11:05, Nick Awesome <jl...@me.com> wrote: > New information, as I said I’m using realtime, > thats the problem! > > I just tested using static config file and it is working perfect. > after some research I figured out that problem with “ps_endpoint_id_ips" for > some reason asterisk ignoring matches in this table, > > I have string in sorcery.conf > > identify = realtime,ps_endpoint_id_ips > > also have string in extconfig.conf > > ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips > > and ofc I have table > > CREATE TABLE `pbx_endpoint_id_ips` ( > `id` varchar(40) NOT NULL, > `endpoint` varchar(40) DEFAULT NULL, > `match` varchar(80) DEFAULT NULL, > UNIQUE KEY `id` (`id`), > KEY `ps_endpoint_id_ips_id` (`id`) > ) ENGINE=InnoDB DEFAULT CHARSET=latin1; > > with entry > > 10001 | 10001 | 85.195.98.178 > > but thats just didn’t works( > > is this a bug and should I open ticket ? > > On 16 Jul 2014, at 21:13, Nick Awesome <jl...@me.com> wrote: > >> 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 <jc...@digium.com> 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:asterisk@85.195.98.178>' failed for >>>> '85.195.98.178:5060' (callid: >>>> 173995aa2e25283807700d65055c9214@85.195.98.178) - 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 > > -- > _____________________________________________________________________ > -- 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
