Try updating your /etc/hosts file with the domain 'testers.com'. Arun
On Sun, May 18, 2014 at 5:06 AM, Olli Heiskanen < ohjelmistoarkkite...@gmail.com> wrote: > Hello, > > It took me a while to get forward on this, but I had progress. I've > changed my sip.conf back and forth so I can't name the exact cause for my > problem, but it may have been the fact that in my asterisk sippeers table > the fields permit and deny may have been in the wrong order. And/or some > configuration values in sip.conf. > > So now clients can register and asterisk 'sip show peer' shows the > registered clients. > > However, there is still one thing that's probably not quite there yet. I'm > using the domain 'testers.com' for my clients, but I can't register them > using that domain. I was able to get clients to register and visible to > asterisk only by using domain '127.0.0.1', if I try commenting that out, > asterisk will say: > chan_sip.c:28073 handle_request_register: Registration from '< > sip:660@127.0.0.1>' failed for '1.1.1.1:5060' - Not a local domain > (where 1.1.1.1 is the public ip of the asterisk+kamailio box) > > In my sip.conf I have domains defined like this: > autodomain=no > domain=127.0.0.1 > domain=testers.com > > I think this may be the cause for this behavior: > In my kamailio.cfg I have asterisk and kamailio bindips defined like this: > asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" > asterisk.bindport = "5070" desc "Asterisk Port" > kamailio.bindip = "127.0.0.1" desc "Kamailio IP Address" > kamailio.bindport = "5060" desc "Kamailio Port" > > And this route forwards REGISTER messages to asterisk using the 127.0.0.1 > as domain: > > route[REGFWD] { > if(!is_method("REGISTER")) > { > return; > } > $var(rip) = $sel(cfg_get.asterisk.bindip); > $uac_req(method)="REGISTER"; > > $uac_req(ruri)="sip:" + $var(rip) + ":" + > $sel(cfg_get.asterisk.bindport); > $uac_req(furi)="sip:" + $au + "@" + $var(rip); > $uac_req(turi)="sip:" + $au + "@" + $var(rip); > $uac_req(hdrs)="Contact: <sip:" + $au + "@" > + $sel(cfg_get.kamailio.bindip) > + ":" + $sel(cfg_get.kamailio.bindport) + > ">\r\n"; > > if($sel(contact.expires) != $null) > $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + > $sel(contact.expires) + "\r\n"; > else > $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + > $hdr(Expires) + "\r\n"; > > uac_req_send(); > } > > > So question is, what would be the good-practice way to fix my setup into > using the client's domain? I thought about using the domain 'testers.com' > in place of kamailio.bindip but was unable to build the sip message and > send it to kamailio ip. uac_req_send() seems to send the message to what is > defined in the request line of the message so replacing it with ' > testers.com' would not work. > > cheers, > Olli > > > > > > > 2014-04-23 17:31 GMT+03:00 Pedro Niño <nino.pe...@gmail.com>: > >> Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your >> sip.conf (asterisk) to show the realtime peers >> El abr 23, 2014 8:29 AM, "Olli Heiskanen" <ohjelmistoarkkite...@gmail.com> >> escribió: >> >> Hello, >>> >>> Gracias Pedro, kiitos Mikko. >>> >>> It's good to know I have configured Kamailio correctly. I added the type >>> into my table but so far no luck having asterisk see the clients >>> registered, at least on cli. I do see that asterisk adds registration data >>> into the table. I'll work on this for a bit and ask in the asterisk list on >>> more tricks on asterisk side. I'll post back here if I find out what the >>> problem was, in case someone is having similar issues. >>> >>> Thanks again, >>> Olli >>> >>> >>> >>> 2014-04-22 21:06 GMT+03:00 Pedro Niño <nino.pe...@gmail.com>: >>> >>>> Don't forget to include peer type (friend), and The callbacknumber In >>>> The table. >>>> >>>> It happened to me and asterisk/kamailio behavior was wayyy to weird >>>> until made sure both parameters were there. >>>> >>>> ----- >>>> >>>> In this setup I have SIP peers in an asterisk table added like this: >>>> >>>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, >>>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' >>>> testers.com'); >>>> >>>> ------ >>>> El abr 19, 2014 1:17 PM, "Olli Heiskanen" < >>>> ohjelmistoarkkite...@gmail.com> escribió: >>>> >>>>> >>>>> Hello, >>>>> >>>>> One of the tests I've been working with is Asterisk realtime >>>>> integration according to Daniel's guide here: >>>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb >>>>> >>>>> Weird thing is the client looks registered but I'm not sure if it >>>>> really is registered. If I'm not mistaken I should see the peers when I >>>>> issue 'sip show peers' on asterisk cli. Instead I get this: >>>>> >>>>> *CLI> sip show peers >>>>> Name/username Host Dyn Forcerport Comedia ACL Port >>>>> Status Description Realtime >>>>> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 >>>>> offline] >>>>> >>>>> >>>>> Also, calling between clients will fail; in Asterisk cli I get: >>>>> *CLI> >>>>> == Using SIP RTP TOS bits 184 >>>>> == Using SIP RTP CoS mark 5 >>>>> -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: >>>>> Dialed 661") in new stack >>>>> -- Executing [661@default:2] Dial("SIP/660-00000000", >>>>> "SIP/661,3600,rt") in new stack >>>>> == Using SIP RTP TOS bits 184 >>>>> == Using SIP RTP CoS mark 5 >>>>> -- Called SIP/661 >>>>> == Everyone is busy/congested at this time (1:0/0/1) >>>>> -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in >>>>> new stack >>>>> == Spawn extension (default, 661, 3) exited non-zero on >>>>> 'SIP/660-00000000' >>>>> >>>>> >>>>> In this setup I have SIP peers in an asterisk table added like this: >>>>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, >>>>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' >>>>> testers.com'); >>>>> >>>>> I have Kamailio and Asterisk on the same machine where Kamailio >>>>> listens port 5060 and Asterisk listens 5070. Things that differ from the >>>>> guide are Kamailio and Asterisk versions, which in my case are newer. >>>>> Also, >>>>> for another testing case I have MULTIDOMAIN enabled in Kamailio, does this >>>>> interfere with the realtime integration? I'm using only one domain though. >>>>> >>>>> Please let me know if any configs or traces I can provide will help >>>>> figure out what's going on. >>>>> >>>>> cheers, >>>>> Olli >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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