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 kamailio...@gmail.com: 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?
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-, 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
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: 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]
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 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-, 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
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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 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-, 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
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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 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-, 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
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Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Olli Heiskanen ohjelmistoarkkite...@gmail.com: 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?
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
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