SIP Clients not unregistering, always qualified

I have installed the latest asterisk server release and using a really simple setup.
I only have 5 clients which can phone each other.
Asterisk is running on a server directly connectet to the internet.
The clients are NATed throught an dynamic ip.

The problem is that when the ip of the clients changes they never get unregistered.
Even with active qualifiying the client are still registered and they can not register again.
The IP is dead for 10 hours and still the status is shown as OK. The new IP hat no open port 5060 or other.

When i restart the server everything works fine again until the next day the ip changes.
I already trying to fix this for a month but it makes no sense. It drives me mad.
Please help.

Name/username Host Dyn Forcerport ACL Port Status Description
66629/66629 D A 5060 OK (82 ms)
66648/66648 D a 21381 OK (168 ms)
66659/66659 D A 6070 OK (145 ms)
66669/66669 D A 5060 OK (79 ms)
66679 (Unspecified) D a 0 UNKNOWN
66689/66689 D A 5060 OK (79 ms)


maxexpiry=600 ; Maximum allowed time of incoming registrations (seconds)
minexpiry=60 ; Minimum length of registrations (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing registration
submaxexpiry=600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry

qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
; and reported in milliseconds with sip show settings.
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
qualifygap=100 ; Number of milliseconds between each group of peers being qualified
; Default: 100
qualifypeers=1 ; Number of peers in a group to be qualified at the same time
; Default: 1
keepalive=60 ; Interval at which keepalive packets should be sent to a peer
; Valid options are yes (60 seconds), no, or the number of seconds.
; Default: 0
nat = auto_force_rport,auto_comedia

ISPs who gratuitously change IP address generally do it to stop you running servers on cheap accounts. Although you refer to them as SIP clients, they are actually servers for calls going to them.

The real solution to this is a better ISP, or a better account with the same ISP.

Your are kidding right?

99 % of all SIP Clients in Germany and Poland are running on dynamic IPs.
Mine was working fine until i updated asterisk to the newstest release due to that secutiry hole.

Although dynamic IP is needed for gratuitous changes, they go one step further and change the IP during established PPP sessions. That has to be done to frustrate servers, as it is basically broken behaviour.

On a well run dynamic system, intended for business use, even different PPP sessions should get the same address, as long as there isn’t too much down time between them. Although I don’t think PPP has specific support for this, DHCP certainly does.

In any case, businesses making any significant use of the internet really should not be cutting corners and using dynamic addresses. I would expect the PPP session to be up 24x7 for any business, so even if they did use dynamic addresses, I would not expect them to change except after network outages.

As a technical point, whilst a SIP user agent is a client when it is registering, when it subsequently receives a call, it is a server.

On the forum you will see that a lot of people have big problems when they are running Asterisk servers that periodically change it’s IP. It looks like Asterisk is just not built for this sort of thing.

Servers by definition do not change their IP’s. That’s just the way it is. If you want to use it in a matter not predicted by the authors, we can not help you. But it is very odd that the setup worked on the previous version. What was that version?

The internet was not built to be used like that!