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@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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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. 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 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 sip%3A100@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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. 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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 -> 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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"); 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 sip%3A200@192.168.10.150>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150 sip%3A100@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.160 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!
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thanks Daniel!
I'm using Debian, so this is helpfull!
thanks again.
2010/9/27 Daniel-Constantin Mierla miconda@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@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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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. 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 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 sip%3A100@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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. 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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 -> 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 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"); 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@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 sip%3A200@192.168.10.150>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150 sip%3A100@192.168.10.150
;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.160 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 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
2010/9/27 Daniel-Constantin Mierla miconda@gmail.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.
Hello,
2010/9/27 Daniel-Constantin Mierlamiconda@gmail.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.
well, it may be useful for others reading the archive in the future :-) .
Cheers, Daniel
-
Daniel-Constantin Mierla -
Iñaki Baz Castillo -
Tincho ylm