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 -------
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 > > >