Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Hello, Thanks for your suggestion, unfortunately it had no effect on the outcome. This (using asterisk-kamailio integration with a domain specified for clients) must have been achieved before, I wonder if I'm doing something wrong here, or is this just not doable? Thanks, Olli 2014-05-18 21:29 GMT+03:00 VOIP Tests : > Try updating your /etc/hosts file with the domain 'testers.com'. > > Arun > > > ___ > 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
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
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: + $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 :
>
>> 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"
>> 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 :
>>>
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 s
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
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);
$uac_req(hdrs)="Contact: \r\n";
if($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 :
> 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"
> 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 :
>>
>>> 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-", "Testing:
Dialed 661") in new stack
-- Executing [661@default:2] Dial("SIP/660-",
"SIP/661,3600,rt") in new stack
== Using SIP RTP TOS bits 184
== Using S
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
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"
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 :
>
>> 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-", "Testing:
>>> Dialed 661") in new stack
>>> -- Executing [661@default:2] Dial("SIP/660-",
>>> "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-", "") in new
>>> stack
>>> == Spawn extension (default, 661, 3) exited non-zero on
>>> 'SIP/660-'
>>>
>>>
>>> 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
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Try to install sngrep In both hosts (Kamailio and asterisk). It will help
you figure what is happening more clear.
I'll double check my own config, and Let you know the needed fields, At
least for my case.
I used the same integration guide, and splitted the model in 3 servers.
One for kamailio, one for databases and one for media server (asterisk).
It's now hosting 350 users with an avg of 15 concurrent calls, planning to
take it to 1200 users in a course of 6 months.
El abr 23, 2014 8:29 AM, "Olli Heiskanen"
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 :
>
>> 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-", "Testing:
>>> Dialed 661") in new stack
>>> -- Executing [661@default:2] Dial("SIP/660-",
>>> "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-", "") in new
>>> stack
>>> == Spawn extension (default, 661, 3) exited non-zero on
>>> 'SIP/660-'
>>>
>>>
>>> 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
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
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 :
> 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"
> 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-", "Testing:
>> Dialed 661") in new stack
>> -- Executing [661@default:2] Dial("SIP/660-",
>> "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-", "") in new
>> stack
>> == Spawn extension (default, 661, 3) exited non-zero on
>> 'SIP/660-'
>>
>>
>> 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
>
>
___
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Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
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"
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-", "Testing:
> Dialed 661") in new stack
> -- Executing [661@default:2] Dial("SIP/660-",
> "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-", "") in new
> stack
> == Spawn extension (default, 661, 3) exited non-zero on
> 'SIP/660-'
>
>
> 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
>
>
___
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Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Olli Heiskanen : > Thanks for the help, here's what I dug up: > > The users are visible in Kamailio, output of kamcmd ul.dump: > (here 1.1.1.1 is the public ip of my Kamailio+Asterisk server and 2.2.2.2 > is the public ip of my home network) Looks like problem is not in Kamailio or SIP message flow. At least I can't spot any problems from registration dance or usrloc sample. -- Mikko Lehto ___ 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
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Hi,
Thanks for the help, here's what I dug up:
The users are visible in Kamailio, output of kamcmd ul.dump:
(here 1.1.1.1 is the public ip of my Kamailio+Asterisk server and 2.2.2.2
is the public ip of my home network)
# kamcmd ul.dump
{
Domain: location
Size: 512
AoRs: {
AoR: 6...@testers.com
HashID: 371821163
Contacts: {
Contact: {
Address: sip:661@192.168.0.106:5062
;transport=udp
Expires: 3588
Q: -1.00
Call-ID: 56c288b61a0baf63@192.168.0.106
CSeq: 20002
User-Agent: Grandstream GXP2000 1.2.5.3
Received: sip:2.2.2.2:5062
Path: [not set]
State: CS_SYNC
Flags: 0
CFlags: 64
Socket: udp:1.1.1.1:5060
Methods: 8159
Ruid: uloc-535276f9-1d83-2
Instance: [not set]
Reg-Id: 0
Last-Keepalive: 1397984187
Last-Modified: 1397984187
}
}
AoR: 6...@testers.com
HashID: 371820875
Contacts: {
Contact: {
Address: sip:660@192.168.0.101:53928
;rinstance=6ca94e284d15eb27;transport=TCP
Expires: 3224
Q: -1.00
Call-ID:
YzQ5YzZlMGI4NTVjNmE3Y2JlMWYzNGI0ODhlMDRmYmI.
CSeq: 12
User-Agent: Z 3.2.21357 r21367
Received: sip:2.2.2.2:1895;transport=TCP
Path: [not set]
State: CS_SYNC
Flags: 0
CFlags: 64
Socket: tcp:1.1.1.1:5060
Methods: 5087
Ruid: uloc-535276f9-1d9e-2
Instance: [not set]
Reg-Id: 0
Last-Keepalive: 1397983823
Last-Modified: 1397983823
}
}
}
Stats: {
Records: 2
Max-Slots: 1
}
}
Yes, the REGISTER message reaches Asterisk and it responds with 200 OK,
here is the SIP trace:
T 2.2.2.2:1895 -> 1.1.1.1:5060 [AP]
REGISTER sip:testers.com;transport=TCP SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.101:53928
;branch=z9hG4bK-d8754z-efb8733fbbad4b7c-1---d8754z-.
Max-Forwards: 70.
Contact: .
To: .
From: ;tag=83057513.
Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY..
CSeq: 1 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
User-Agent: Z 3.2.21357 r21367.
Allow-Events: presence, kpml.
Content-Length: 0.
.
T 1.1.1.1:5060 -> 2.2.2.2:1895 [AP]
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/TCP 192.168.0.101:53928
;branch=z9hG4bK-d8754z-efb8733fbbad4b7c-1---d8754z-;rport=1895;received=2.2.2.2.
To: ;tag=9bf08f91615a4a194285ad1308f058f3.c9e3.
From: ;tag=83057513.
Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY..
CSeq: 1 REGISTER.
WWW-Authenticate: Digest realm="testers.com",
nonce="U1OO1VNTjalLQmkJEOAWOaLqbTEALvG7".
Content-Length: 0.
.
T 2.2.2.2:1895 -> 1.1.1.1:5060 [AP]
REGISTER sip:testers.com;transport=TCP SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.101:53928
;branch=z9hG4bK-d8754z-2433219a229ba04d-1---d8754z-.
Max-Forwards: 70.
Contact: .
To: .
From: ;tag=83057513.
Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY..
CSeq: 2 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
User-Agent: Z 3.2.21357 r21367.
Authorization: Digest username="660",realm="testers.com
",nonce="U1OO1VNTjalLQmkJEOAWOaLqbTEALvG7",uri="sip:testers.com
;transport=TCP",response="ca517e546b71ac50d94fe34032d36e7d",algorithm=MD5.
Allow-Events: presence, kpml.
Content-Length: 0.
.
T 1.1.1.1:5060 -> 2.2.2.2:1895 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.101:53928
;branch=z9hG4bK-d8754z-2433219a229ba04d-1---d8754z-;rport=1895;received=2.2.2.2.
To: ;tag=9bf08f91615a4a194285ad1308f058f3.9743.
From: ;tag=83057513.
Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY..
CSeq: 2 REGISTER.
Contact: ;expires=3600;received="sip:2.2.2.2:1895
;transport=TCP".
Content-Length: 0.
.
U 1
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
2014-04-19 (Sat) 20:46 UTC +0300 Olli Heiskanen : > 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: Hi Olli Some pointers: Can you see users on Kamailio usrloc? Can you see REGISTER requests arriving to Asterisk? Are there any errors on Asterisk when REGISTER request is handled? You should add some xlog() to AUTH, REGISTRAR and REGFWD routes in Kamailio. Also, maybe you could provide these outputs: kamcmd ul.lookup location 6...@testers.com kamcmd ul.lookup location 6...@testers.com kamcmd ul.dump ngrep -d any -W byline -q port 5060 or port 5070 Regards -- Mikko Lehto ___ 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
