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
...@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 mov
On Mar 22, 2013, at 5:22 AM, Florian Wolters 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 b
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 exam
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 the
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
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 p
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 t
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 la
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 wrote:
> Hi @ll,
>
> I just moved my Asterisk Box and changed the Provider and Internet Access
> to a full IP Ac
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 mov
2013/3/21 Florian Wolters :
> 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 proble
Try canreinvite=yes in sip trunk
-Original Message-
From: Florian Wolters
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk disconnecting SIP
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 inc
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