27 Sep
2010
27 Sep
'10
11:31 a.m.
btw, if you want to install from sources, here is a tutorial for 3.0.x:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git
If you work with debian or ubuntu, there are apt repos for them:
http://www.kamailio.org/dokuwiki/doku.php/packages:debs
Cheers,
Daniel
Hello,
the r-uri is not rewritten with ip address of the phone, I guess you
don't use user location to locate the phone. Is the phone registered
to kamailio?
You say about the code for re-invites where you have a t_relay with
outbound proxy. Normally, that should go via record-routing. If that
code is also for initial invites and you must do it in this way, then
you need to rewrite the r-uri domain and port to match phone's ip and
port.
I suggest you use kamailio 3.0.x with default config file. It is easy
to enable features such as authentication and use location. Create
accounts for you phones, set them to register to kamailio and make
calls. Then adapt the config to meet extra needs you may have.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi all!
I really don't know why "Mitel" rejects my calls. I'm using Kamailio
to forward calls to Mitel.
A little more graphic:
Please see the picture:
http://s3.subirimagenes.com:81/otros/5226539form.jpg
SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone
Mitel rejects my calls with "404 Not Found". Ok, you may think: "the
extension that you are calling doesn't exists".. please dont think that.
(One more thing: If I try to make the same scene using Asterisk
instead Kamailio everything works fine.)
So, I made a sip capture to see what happens:
Sip Phone -> 100
192.168.10.140 -> Sip Phone
192.168.10.150 -> Kamailio
192.168.10.160 -> Mitel
Mitel Phone -> 200
Kamailio
U 192.168.10.140:5060 <http://192.168.10.140:5060> ->
192.168.10.150:5060 <http://192.168.10.150:5060>
INVITE sip:200@192.168.10.150 <mailto:sip%3A200@192.168.10.150> SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at]
with a @).
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: "Sip Phone" <sip:100@192.168.10.140:5060
<http://sip:100@192.168.10.140:5060>>.
Expires: 240.
User-Agent: Linksys/SPA941-5.1.8.
Content-Length: 395.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: replaces.
Content-Type: application/sdp.
U 192.168.10.150:5060 <http://192.168.10.150:5060> ->
192.168.10.140:5060 <http://192.168.10.140:5060>
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.10.140:5060;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140.
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at]
with a @).
CSeq: 101 INVITE.
Server: Kamailio (1.5.4-notls (i386/linux)).
Content-Length: 0.
U 192.168.10.150:5060 <http://192.168.10.150:5060> ->
192.168.10.160:5060 <http://192.168.10.160:5060>
INVITE sip:200@192.168.10.150 <mailto:sip%3A200@192.168.10.150> SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0.
Via: SIP/2.0/UDP
192.168.10.140:5060;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at]
with a @).
CSeq: 101 INVITE.
Max-Forwards: 69.
Contact: "Sip Phone" <sip:100@192.168.10.140:5060
<http://sip:100@192.168.10.140:5060>>.
Expires: 240.
User-Agent: Linksys/SPA941-5.1.8.
Content-Length: 395.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: replaces.
Content-Type: application/sdp.
U 192.168.10.160:5060 <http://192.168.10.160:5060> ->
192.168.10.150:5060 <http://192.168.10.150:5060>
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP
192.168.10.140:5060;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>;tag=0_4044193584-65506210.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at]
with a @).
CSeq: 101 INVITE.
Content-Length: 0.
U 192.168.10.160:5060 <http://192.168.10.160:5060> ->
192.168.10.150:5060 <http://192.168.10.150:5060>
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP
192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP
192.168.10.140:5060;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>;tag=0_4044193584-65506210.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at]
with a @).
CSeq: 101 INVITE.
Contact: <sip:192.168.10.160>.
Content-Length: 0.
This is my Kamailio code from reenvites..
route[4] {
t_relay("udp:192.168.10.160:5060 <http://192.168.10.160:5060>");
t_on_reply("1");
exit;
}
If you pay attention to INVITES (Kamailio SIP messages) you will see:
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150
<mailto:sip%3A200@192.168.10.150>>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150
<mailto:sip%3A100@192.168.10.150>>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.160
<mailto:sip%3A200@192.168.10.160>>.
It could be the reason for Mitel rejects? Can I fix it? I can use
TEXTOPS but I cant understand why Mitel rejects the Kamailio INVITES.
I will thanks any help!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 