[SR-Users] Realtime integration: Unregistered clients showing as registered?
VOIP Tests
kamailio.fs at gmail.com
Sun May 18 20:29:49 CEST 2014
Try updating your /etc/hosts file with the domain 'testers.com'.
Arun
On Sun, May 18, 2014 at 5:06 AM, Olli Heiskanen <
ohjelmistoarkkitehti at 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 at 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.pedro at 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" <ohjelmistoarkkitehti at 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.pedro at 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" <
>>>> ohjelmistoarkkitehti at 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 at default:1] NoOp("SIP/660-00000000", "Testing:
>>>>> Dialed 661") in new stack
>>>>> -- Executing [661 at 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 at 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 at 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 at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>> _______________________________________________
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>>> sr-users at lists.sip-router.org
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>>>
>> _______________________________________________
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>>
>>
>
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>
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