[Asterisk-Users] Linksys register hangs Asterisk!

[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  -------