<div dir="ltr">hi,<div>it seems your vpn connection drop.</div><div>is you vpn on WiFi of any other high latency network? </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI <span dir="ltr"><<a href="mailto:admin@tootai.net" target="_blank">admin@tootai.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem.<br>
<br>
>From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s<br>
<br>
In logs I see<br>
<br>
WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See <a href="https://wiki.a" target="_blank">https://wiki.a</a><br>
Packet timed out after 6401ms with no response (or 6399ms or 6401ms)<br>
<br>
Qualify freq being 60000 ms for the peers.<br>
<br>
For the SIP peers I see<br>
<br>
Sess-Timers : Accept<br>
Sess-Refresh : uas<br>
Sess-Expires : 1800 secs<br>
Min-Sess : 90 secs<br>
<br>
I tried to use originate for session-timer in global SIP conf, no changes.<br>
<br>
Any hint about this matter would be appreciate.<span class="HOEnZb"><font color="#888888"><br>
<br>
-- <br>
Daniel<br>
<br>
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