23 Sep
2010
23 Sep
'10
3 p.m.
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 -> 192.168.10.150:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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>.
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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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 -> 192.168.10.160:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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>.
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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)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");
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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>0>>.
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!
27 Sep
27 Sep
12:29 p.m.
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
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
12:31 p.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
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8:39 p.m.
New subject: [SR-Users] Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?
Thanks Daniel!
I'm using Debian, so this is helpfull!
thanks again.
2010/9/27 Daniel-Constantin Mierla <miconda(a)gmail.com>
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
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 -> 192.168.10.150:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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>.
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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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 -> 192.168.10.160:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
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>.
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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)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 -> 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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)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");
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 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>0>>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
;tag=d396005aaf3ab9a2o0.
To: "Mitel
Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>0>>.
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
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://www.asipto.com
5:52 p.m.
New subject: [SR-Users] Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?
2010/9/27 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
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?
Hi Daniel, please don't waste your time as this user already asked the
same question in sr-users-es maillist, and I already replied him:
http://lists.sip-router.org/pipermail/sr-users-es/2010-September/004303.html
He just ignored my response and has opened the same thread here.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
6:08 p.m.
Hello,
2010/9/27 Daniel-Constantin
Mierla<miconda(a)gmail.com>om>:
well, it may be useful for others reading the archive in the future :-) .
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
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?
Hi
Daniel, please don't waste your time as this user already asked the
same question in sr-users-es maillist, and I already replied him:
http://lists.sip-router.org/pipermail/sr-users-es/2010-September/004303.html
He just ignored my response and has opened the same thread here.
5170
days inactive
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5 comments
3 participants
participants (3)
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Daniel-Constantin Mierla -
Iñaki Baz Castillo -
Tincho ylm