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

[Darren Nickerson]

On Mar 22, 2013, at 5:22 AM, Florian Wolters <florian at florian-wolters.de> wrote:
> 
> So I did setup another Extension leading me to a MeetMe conference to at
> least listen to some MoH while waiting for the 15 Minutes to exceed. This
> showed the same behaviour. After exactly 15 Minutes, the call is
> terminated  - namely by the provider. The analysis of the dump in
> Wireshark shows the last 6 SIP packets:
> 
> 2013-03-21 15:56:50.648141    217.0.17.170   =>   172.16.0.2    Request:
> INVITE sip:02341234567890 at 79.253.136.186:5060
> 2013-03-21 15:56:50.648325    172.16.0.2     =>   217.0.17.170  Status:
> 100 Trying
> 2013-03-21 15:56:50.648427    172.16.0.2     =>   217.0.17.170  Status:
> 200 OK, with session description
> 2013-03-21 15:56:50.731436    217.0.17.170   =>   172.16.0.2    Request:
> ACK sip:02341234567890 at 79.253.136.186:5060
> 2013-03-21 15:56:50.735426    217.0.17.170   =>   172.16.0.2    Request:
> BYE sip:02341234567890 at 79.253.136.186:5060
> 2013-03-21 15:56:50.735590    172.16.0.2     =>   217.0.17.170  Status:
> 200 OK
> 
> (manually copied that from the Wireshark window). This looks to me as if
> the provider for some reason does an INVITE after 15 Minutes, that is not
> correctly handled by my Asterisk. Is there any timer inside the SIP
> protocol, that may be aged by 15 Minutes? Or should I have a deeper look
> at the SIP packets?

Sip session timers? 

http://doxygen.asterisk.org/trunk/sip_session_timers.html

-d