[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[Jim Lucas]

On 3/21/2013 12:31 AM, Florian Wolters wrote:
> Hi @ll,
>
> I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom.
>
> I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working.
>
> The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here.
>
> I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success.
>
> Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential.
>
> Best regards
>
>     Flo

Florian,

As both an VoIP provider and phone system vendor, I had this same 
problem 2 years ago.  In my situation, it turned out that it was nothing 
to do with either the Asterisk box or the provider.

The problem was with a router that we had terminating our T1 connection. 
  As an ISP we provide T1's to many customers and we provide the router 
as well.  In this specific case, the customer purchased a data T1 
connection with QoS (sip and rtp) then purchased our IP asterisk phone 
system with SIP trunks from us as well.

The way we found this issue was by switching our the T1 router.  Turns 
out that it fixed the problem.  Exact same configuration was on each 
router.  So we started scratching our heads...

We then looked at the firmware of the two routers and found that they 
were different.

We provide Cisco 26XX routers.

Their are many places on the net talking about the 15 minute NAT timeout 
issue.

If you are not using this device, well, maybe it has a similar bug.

--
Jim Lucas