Here is a update with the solution.. Reinstallation of Debian! I think it was an update of Debian Unstable that made things stop working. Now I installed Debian stable with the same config and it works great now.
Even that noone replied to my post thanks for reading it anyway! =) ~Johannes > Hey, > > I'w got a problem (bug maybe?). > > I have recently got my Asterisk to work perfect and I'm not trying to > setup some dial routes and get the system working as I wan't it to. > > Yesterday I was installing Festival and also did a "aptitude upgrade" on > my Debian Unstable installation. > After that the problem started. > > After some serious testing yesterday night and today I have tracked down > the problem to that it it is my Linksys WRTG54GP2 (Router with ATA) that > causes asterisk to stop working. > > Everytime it tries to register asterisk stops working normally. It don't > register any more information with sip debug activated. No incoming calls > is displayed and asterisk seems just to be seeing nothing that is going > on. > > I tried to restart asterisk and then make a incoming call directly, that > goes well. Asterisk answers and posts the normal route with voice answers. > Then I can see that the Linksys router is trying to register and after > that everything stops working. > > If I disable the linksys router to register itself everything works well, > asterisk answers and gived me the options to choose extension. > > So the problem is caused by the registration of Linksys. > This is the debug log from the registration until asterisk stops (moved to > the bottom of this mail) > > One interesting line is that the "Call-ID:" line after the @ contains the > IP number to the Linksys router WITHOUT THE LAST NUMBER in the address! > How can that be? The other lines containg the IP number is correct (in the > log replaced by <Linksys-IP>). > Can this be the cause for the problem ? > If not can there be anything else in this log that indicates what the > problem is? > > Hope someone got an answer because this is driving me crazy since I got it > all working this weekend after 2 weeks of trouble. > > Regards, > ~Johannes > > ------ START SIP DEBUG LOG ------- > Sip read: > REGISTER sip:<server-IP> SIP/2.0 > Via: SIP/2.0/UDP <Linksys-IP>:5060;branch=z9hG4bK-d54de2e6 > From: <sip:100@<server-IP>>;tag=4c7b1b149bb4b329o0 > To: <sip:100@<server-IP>> > Call-ID: 66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST DIGIT IN NUMBER> > CSeq: 1 REGISTER > Max-Forwards: 70 > Contact: <sip:100@<Linksys-IP>:5060>;expires=3600 > User-Agent: Linksys/RT31P2-3.1.3(LI) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > > > 12 headers, 0 lines > Using latest request as basis request > Sending to <Linksys-IP> : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP <Linksys-IP>:5060;branch=z9hG4bK-d54de2e6 > From: <sip:100@<server-IP>>;tag=4c7b1b149bb4b329o0 > To: <sip:100@<server-IP>> > Call-ID: 66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST DIGIT IN NUMBER> > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:100@<server-IP>> > Content-Length: 0 > > > to <Linksys-IP>:5060 > Transmitting (no NAT): > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP <Linksys-IP>:5060;branch=z9hG4bK-d54de2e6 > From: <sip:100@<server-IP>>;tag=4c7b1b149bb4b329o0 > To: <sip:100@<server-IP>>;tag=as7ba88dca > Call-ID: 66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST DIGIT IN NUMBER> > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:100@<server-IP>> > WWW-Authenticate: Digest realm="asterisk", nonce="7b426d2d" > Content-Length: 0 > > > to <Linksys-IP>:5060 > Scheduling destruction of call '66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST > DIGIT IN NUMBER>' in 15000 ms > debian*CLI> > > Sip read: > REGISTER sip:<server-IP> SIP/2.0 > Via: SIP/2.0/UDP <Linksys-IP>:5060;branch=z9hG4bK-2d99db8a > From: <sip:100@<server-IP>>;tag=4c7b1b149bb4b329o0 > To: <sip:100@<server-IP>> > Call-ID: 66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST DIGIT IN NUMBER> > CSeq: 2 REGISTER > Max-Forwards: 70 > Authorization: Digest > username="100",realm="asterisk",nonce="7b426d2d",uri="sip:<server-IP>",algorithm=MD5,response="b904 > 95eaf088d8696ac0cc5ebad9f990" > Contact: <sip:100@<Linksys-IP>:5060>;expires=3600 > User-Agent: Linksys/RT31P2-3.1.3(LI) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > > 13 headers, 0 lines > Using latest request as basis request > Sending to <Linksys-IP> : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP <Linksys-IP>:5060;branch=z9hG4bK-2d99db8a > From: <sip:100@<server-IP>>;tag=4c7b1b149bb4b329o0 > To: <sip:100@<server-IP>> > Call-ID: 66a3a900-ec8d9e2d@<Linksys-IP MISSING LAST DIGIT IN NUMBER> > CSeq: 2 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:100@<server-IP>> > Content-Length: 0 > > > to <Linksys-IP>:5060 > ------ STOP ------- > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
