Jim,
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.
As I am using a fli4l (Linux Router), this seems to not be the problem. I
cannot see any dropped packets or timeouts in the logfiles of
Matthew and list,
thanks for your detailed reply.
This is a little hard to diagnose without seeing the SIP traffic for the
duration of the call. It makes it impossible to tell if the INVITES the
provider is sending are related to the call (i.e. have the same Call-ID
header),
but if they
Hi List,
Try canreinvite=yes in sip trunk
This did not make any difference... -.-
-Original Message-
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
On Mar 22, 2013, at 5:22 AM, Florian Wolters flor...@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
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Florian Wolters
Sent: Thursday, March 21, 2013 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
Hello,
I solved it by moving
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
Try canreinvite=yes in sip trunk
-Original Message-
From: Florian Wolters flor...@florian-wolters.de
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
2013/3/21 Florian Wolters flor...@florian-wolters.de:
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
I had this exact problem with my voip provider a few years ago.
It was disconnecting at exactly 5 minutes.
I solved it by moving Asterisk 1.6 to Asterisk 1.4.
Try asterisk 1.4 or 1.8 on a test box and see how it goes.
Peter
On 21/03/2013 09:31, Florian Wolters wrote:
Hi @ll,
I just moved
I am having the same problem with Asterisk 11.2.0 and Linphone and it is
exactly 15 minutes and occurring with SIP running on our LAN.
On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de
wrote:
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet
Hello,
I solved it by moving Asterisk 1.6 to Asterisk 1.4.
Try asterisk 1.4 or 1.8 on a test box and see how it goes.
I did try the latest 1.8.2x release already without any improvement.
Currently running is a Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 as the tcpdump
says (little mistake to my last
Florian Wolters wrote:
So I turned on SIP debug for this host and analyszed it with wireshark.
The last packets show an INVITE from my provider, that is answered by my
Asterisk with 200 OK, with session description. What follows is an ACK
by the provider and immediately a BYE sent by the
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
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