<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 style>Here is the message</div><div style><br></div><div><div>Reliably Transmitting (no NAT) to <a href="http://10.200.1.55:5076">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">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">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">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">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 style>
Please note, I have to stick with TCP because of packet loss in the network </div><div style><br></div><div>Any suggestions? </div><div><br clear="all"><div><div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div></div>
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