<div dir="ltr"><div>;ignoreregexpire=yes ; Enabling this setting has two functions:<br> ;<br> ; For non-realtime peers, when their registration expires, the<br>
; information will _not_ be removed from memory or the Asterisk database<br> ; if you attempt to place a call to the peer, the existing information<br> ; will be used in spite of it having expired<br>
;<br> ; For realtime peers, when the peer is retrieved from realtime storage,<br> ; the registration information will be used regardless of whether<br>
; it has expired or not; if it expires while the realtime peer<br> ; is still in memory (due to caching or other reasons), the<br> ; information will not be removed from realtime storage<br>
</div><div>Also remove all qualify related parameters and keepalive if set</div><div> </div><div>Hope it will solve your problem</div><div> </div><div>Regards,</div><div> </div><div>Bharat Lalcheta<br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Apr 16, 2013 at 11:26 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"><div dir="ltr"><div>Here is what I have, also attached sip show settings output and part of sip.conf in issues</div><div><br></div><div>[general]</div><div>udpbindaddr=<a href="tel:172.20.255.40" target="_blank" value="+911722025540">172.20.255.40</a></div>
<div>transport=udp,tcp</div>
<div>tcpenable=yes</div><div>tlsenable=no</div><div>tcpbindaddr=<a href="tel:172.20.255.40" target="_blank" value="+911722025540">172.20.255.40</a></div><div>directrtpsetup=no</div><div>directmedia=yes</div><div>allowguest=no</div>
<div>match_auth_username=yes</div><div>tos_sip=AF31</div><div>
tos_audio=ef</div><div>tos=0xB8</div><div>tos_video=af41 ; Sets TOS for RTP video packets.</div><div>tos_text=af41 ; Sets TOS for RTP text packets.</div><div>trustrpid = yes ; If Remote-Party-ID should be trusted</div>
<div>sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)</div><div>disallow=all</div><div>allow=alaw</div><div>allow=ulaw</div><div>allow=g729</div><div>maxforwards=70</div><div>relaxdtmf=yes</div>
<div>rpid_update = yes</div><div>maxexpiry=400</div><div>minexpiry=60</div><div>defaultexpiry=300</div><div>qualify=yes ;</div><div>notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones)</div>
<div>qualifyfreq=300</div><div>qualifypeers=1</div><div>qualifygap=2000</div><div>registertimeout=20</div><div>registerattempts=10</div><div>progressinband=never</div><div>ignoreregexpire=yes</div><div><div class="h5"><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta <span dir="ltr"><<a href="mailto:bharatlalcheta@gmail.com" target="_blank">bharatlalcheta@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid">
<div dir="ltr"><div>Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario.</div><div> </div><div>Regards,</div><div> </div><div>Bharat Lalcheta</div><div> </div></div>
<div class="gmail_extra"><div><div><br><br><div class="gmail_quote">On Tue, Apr 16, 2013 at 11:03 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:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><div dir="ltr">Backtrace and logs attached here : <a href="https://issues.asterisk.org/jira/browse/ASTERISK-21447" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-21447</a><br clear="all">
<div><div dir="ltr"><br></div><div dir="ltr">
Regards,<br>Zohair Raza<div><br></div><div><div> <br></div></div></div></div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry <span dir="ltr"><<a href="mailto:markhenry430@gmail.com" target="_blank">markhenry430@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><div dir="ltr">this is my secondary email <div><br></div>
<div>Regards</div><div>Zohair</div></div><div>
<div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <span dir="ltr"><<a href="mailto:markhenry430@gmail.com" target="_blank">markhenry430@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><div dir="ltr">Tried disabling qualify and changing frequency with qualify=yes already, no luck :(<div>
<div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf <span dir="ltr"><<a href="mailto:mehroz.ashraf85@gmail.com" target="_blank">mehroz.ashraf85@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><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"><div><div>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>
</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><div><div><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 <a href="tel:172.20.255.50" target="_blank" value="+911722025550">172.20.255.50</a>: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@<a href="tel:172.20.255.50" target="_blank" value="+911722025550">172.20.255.50</a>: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><span><font color="#888888">
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