[Asterisk-Users] Linksys register hangs Asterisk!

[Johannes]

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  -------
>
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