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

[Matthew J. Roth]

Florian Wolters wrote:
> 
> Does it make sense to have a more detailed tcpdump of the SIP session? If
> so, how should such a thing been shared without posting too much ASCII
> text to the list?

SIP sessions are generally small enough to post right to the list.  Otherwise,
you can put them up on a site like pastebin.com and provide the link.

> 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?

This is where a full SIP trace that includes the messages used to setup the call
in the first place would be helpful.  I haven't seen anything related to session
timers in what you've posted so far, but they may have been negotiated when the
call was established.

Regardless, your calls are consistently dropping at 15 minutes and you've shown
that it's caused by the provider sending an INVITE, waiting for the OK, and
then sending a BYE.  You have enough to go to them and ask why it's happening.
Even if it's something in your Asterisk configuration, they are initiating the
hangup and should be able to tell you why.  If they can't or won't help you
troubleshoot this problem then I'd seriously consider looking for a new
provider.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer