Thanks Daniel! I'm using Debian, so this is helpfull!
thanks again. 2010/9/27 Daniel-Constantin Mierla <mico...@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:2...@192.168.10.150 <sip%3a...@192.168.10.150> SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. > From: "Sip Phone" <sip:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@192.168.10.150>>. > > I think that should be: > > From: "Sip Phone" <sip:1...@192.168.10.150 <sip%3a...@192.168.10.150> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:2...@192.168.10.160 <sip%3a...@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-us...@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-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.com > >
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