18 May
2014
18 May
'14
9:29 p.m.
Try updating your /etc/hosts file with the domain 'testers.com'.
Arun
On Sun, May 18, 2014 at 5:06 AM, Olli Heiskanen <
ohjelmistoarkkitehti(a)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.pedro(a)gmail.com>om>:
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(a)gmail.com>
escribió:
Hello,
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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(a)gmail.com>om>:
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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(a)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
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>
>
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