<div dir="ltr">I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when "qualify" are not acknowledged. You can also check "qualifyfreq" to limit the number of qualifies for particular peer. </div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza <span dir="ltr"><<a href="mailto:engineerzuhairraza@gmail.com" target="_blank">engineerzuhairraza@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello List, <div><br></div><div>Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received </div>
<div><br></div><div>It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck </div>
<div><br></div><div>Here is the message</div><div><br></div><div><div>Reliably Transmitting (no NAT) to <a href="http://10.200.1.55:5076" target="_blank">10.200.1.55:5076</a>:</div><div>OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0</div>
<div>Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd</div><div>Max-Forwards: 70</div><div>From: "Unknown" <<a href="mailto:sip%3AUnknown@172.20.255.50" target="_blank">sip:Unknown@172.20.255.50</a>>;tag=as6c5371b0</div>
<div>To: <sip:2271@10.200.1.55:5076;transport=tcp></div><div>Contact: <sip:Unknown@172.20.255.50:5060;transport=TCP></div><div>Call-ID: <a href="http://433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060" target="_blank">433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060</a></div>
<div>CSeq: 101 OPTIONS</div><div>User-Agent: ASTPBX</div><div>Date: Mon, 15 Apr 2013 15:25:09 GMT</div><div>Session-Expires: 80</div><div>Min-SE: 90</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div>
<div>Supported: replaces, timer</div><div>Content-Length: 0</div><div><br></div><div><br></div><div>---</div><div>[2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to <a href="http://10.200.1.55:5076" target="_blank">10.200.1.55:5076</a> returned -2: Interrupted syste</div>
</div><div><br></div><div>Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message </div><div><br></div><div><div>[2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to <a href="http://10.200.1.55:5075" target="_blank">10.200.1.55:5075</a>: Connection refused</div>
<div>[2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded.</div></div><div><br></div><div>I will produce a back trace later today and file a bug, I am using version 1.8.14.0 <br></div><div><br></div><div>
Please note, I have to stick with TCP because of packet loss in the network </div><div><br></div><div>Any suggestions? </div><div><br clear="all"><div><div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div></div>
</div></div>
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