Internal lines changed to unavailable state after ~1 hour

Hello all, I’m hoping some of the wisdom of the masses here can help me out. We’ve got a asterisk based trixbox system which has developed a strange problem…

After we reset our phones, they successfully register with the system however after about an hour even if NO other activity occurs (ie, this message occurs directly after the line registration message) this happens:

Extension Changed 100 new state Unavailable for Notify User 100
Extension Changed 100 new state Unavailable for Notify User 101
Extension Changed 101 new state Unavailable for Notify User 100
Extension Changed 101 new state Unavailable for Notify User 101

We’re at a loss here, then both phones show the lines unavailable and calls do not come in…

Thank you in advance for any ideas, the phones are Grandstream GXP-2000’s if it’s relevant.

-Nic

Did you try

:question:

I just activated the debugging (took me a bit to figure out the command is “sip debug peer” on my system). I’ll monitor the output and see if anything strange occurs.

Thanks for the suggestion!

This occurs every 30 seconds or so, it looks like an unknown incoming call? It comes in and then is destroyed 5-10 seconds later…

10.0.0.118 is my phone. 10.0.0.5 is our trixbox.

[code]—
trixbox*CLI>
<-- SIP read from 10.0.0.118:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK7f3304e4;rport
From: “Unknown” sip:Unknown@10.0.0.5;tag=as2450ee7a
To: sip:101@10.0.0.118:5060;transport=udp;tag=as6a56eaba
Call-ID: 5acc652846b6a77314fc598047a94da8@10.0.0.5
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: sip:101@10.0.0.118:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0

— (11 headers 0 lines) —
Destroying call ‘5acc652846b6a77314fc598047a94da8@10.0.0.5’
[/code]

Could this be related to the line unavailability?

Okay some more output from the debug surrounding the lines going unavailable.

---
Reliably Transmitting (NAT) to 10.0.0.118:5060:
NOTIFY sip:101@10.0.0.118:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5fd0b84c;rport
From: <sip:300@10.0.0.5>;tag=as1b0991b6
To: "Nic Orrben" <sip:101@10.0.0.5>;tag=91c0e79998993ac0
Contact: <sip:300@10.0.0.5>
Call-ID: 87a158513f948dda@10.0.0.118
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 199

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="full" entity="sip:300@10.0.0.5">
<dialog id="300">
<state>terminated</state>
</dialog>
</dialog-info>

---
trixbox*CLI>
<-- SIP read from 10.0.0.118:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5fd0b84c;rport
From: <sip:300@10.0.0.5>;tag=as1b0991b6
To: "Nic Orrben" <sip:101@10.0.0.5>;tag=91c0e79998993ac0
Call-ID: 87a158513f948dda@10.0.0.118
CSeq: 104 NOTIFY
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: <sip:101@10.0.0.118:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


--- (11 headers 0 lines) ---
Reliably Transmitting (NAT) to 10.0.0.118:5060:
NOTIFY sip:101@10.0.0.118:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK13884af3;rport
From: <sip:101@10.0.0.5>;tag=as736d2d40
To: "Nic Orrben" <sip:101@10.0.0.5>;tag=f785187f4fec1a76
Contact: <sip:101@10.0.0.5>
Call-ID: 7331c9f999f95931@10.0.0.118
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 198

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="9" state="full" entity="sip:101@10.0.0.5">
<dialog id="101">
<state>confirmed</state>
</dialog>
</dialog-info>

---
 Extension Changed 101 new state Unavailable for Notify User 101
 Extension Changed 101 new state Unavailable for Notify User 100
trixbox*CLI>
<-- SIP read from 10.0.0.118:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK13884af3;rport
From: <sip:101@10.0.0.5>;tag=as736d2d40
To: "Nic Orrben" <sip:101@10.0.0.5>;tag=f785187f4fec1a76
Call-ID: 7331c9f999f95931@10.0.0.118
CSeq: 111 NOTIFY
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: <sip:101@10.0.0.118:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0

Sorry I’m not experienced enough with the system to parse what this is telling me…

:confused:

just smthng to try: add localnet=10.0.0.0/255.0.0.0 to your sip.conf

( http://www.voip-info.org/wiki/view/Asterisk+SIP+localnet )