<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style></head>
<body class='hmmessage'><div dir='ltr'>Hi George,<BR> <BR>We need to store contacts in realtime for our system. However not all endpoints are registered only about 200, yet asterisk loops through every endpoint which has been defined. It does this if contacts are in realtime or not.<BR> <BR>Its almost like pjsip is loading them to check if they need to be qualified etc.<BR> <BR>Asterisk 1.8 only put things into cache once they were accessed, is this an option for sourcery?<BR> <BR>Thanks,<BR> <BR>Ross<br> <BR><div><hr id="stopSpelling">From: george.joseph@fairview5.com<br>Date: Tue, 1 Mar 2016 10:42:58 -0700<br>To: asterisk-dev@lists.digium.com<br>Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241<br><br><div dir="ltr"><div class="ecxgmail_default" style="font-family: arial narrow,sans-serif;"><br></div><div class="ecxgmail_extra"><br><div class="ecxgmail_quote">On Tue, Mar 1, 2016 at 10:29 AM, Ross Beer <span dir="ltr"><<a href="mailto:ross.beer@outlook.com" target="_blank">ross.beer@outlook.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;">
<div><div dir="ltr">Hi George,<br> <br>I have now got asterisk 13 trunk, however loading is very slow. This is due to asterisk reading all of the realtime Sorcery peers and marking them all as 'Unknown'. Is there a way to only cache peers that have tried to register?<br></div></div></blockquote><div><br></div><div><br></div><div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif; display: inline;'>When you say "Asterisk 13 trunk"</div> <div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif; display: inline;'>you do mean "branch" correct?</div></div><div><br></div><div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif;'>Assuming you have contacts coming from realtime, the only was to prevent them from being qualified is to delete them from the ps_contacts table before starting Asterisk. You really don't gain anything by using realtime for contacts anyway. I'd just disable it and let Asterisk use the internal sqlite3 database to keep track of them.</div></div><div> <br></div><blockquote class="ecxgmail_quote" style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;"><div><div dir="ltr"> <br>So far its taking 20 mins to load!!<br> <br>Also asterisk has the following warning:<br> <br>taskprocessor.c:803 taskprocessor_push: The 'subm:ast_device_state_topic-000055d0' task processor queue reached 500 scheduled tasks.<br> <br></div></div></blockquote><div><br></div><div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif;'>Whoa! This makes me think I might have messed something up in the fix for contacts not being cached correctly.</div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif;'><br></div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif;'>Don't use realtime for contacts and see what happens. I'm going to re-test.</div><div class="ecxgmail_default" style='font-family: "arial narrow",sans-serif;'><br></div></div><div> </div><blockquote class="ecxgmail_quote" style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;"><div><div dir="ltr">Neither were issues in the previous release.<br> <br>Thank you for your assistance,<br> <br>Ross<br> <br><div><hr>From: <a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a><br>Date: Tue, 1 Mar 2016 09:08:37 -0700<br>To: <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><div><div class="h5"><br>Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241<br><br><div dir="ltr"><div style="font-family: arial narrow,sans-serif;"><br></div><div><br><div>On Tue, Mar 1, 2016 at 5:58 AM, Ross Beer <span dir="ltr"><<a href="mailto:ross.beer@outlook.com" target="_blank">ross.beer@outlook.com</a>></span> wrote:<br><blockquote style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;">
<div><div dir="ltr">Further to my previous email it appears this bug won't be easily resolved by changing the method:<br> <br>pjsip_dlg_create_uas() >> pjsip_dlg_create_uas_and_inc_lock().<br> <br>Asterisk starts ok, allows registrations but no calls progress.<br></div></div></blockquote><div><br></div><div><br></div><div><div style='font-family: "arial narrow",sans-serif;'>You have to pull Asterisk from the 13 branch. This should have been fixed with review 2236 and I've been running with that patch and pjproject trunk.</div><div style='font-family: "arial narrow",sans-serif;'><br></div></div><div> <br></div><blockquote style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;"><div><div dir="ltr"> <br><div><span><hr>From: <a href="mailto:ross.beer@outlook.com" target="_blank">ross.beer@outlook.com</a><br>To: <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><br></span>Date: Tue, 1 Mar 2016 11:49:55 +0000<br>Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241<div><div><br><br>
<div dir="ltr">I've just found an open issue for this <a href="https://issues.asterisk.org/jira/browse/ASTERISK-25751" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-25751</a><br><br> <br><div><hr>From: <a href="mailto:ross.beer@outlook.com" target="_blank">ross.beer@outlook.com</a><br>To: <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><br>Date: Tue, 1 Mar 2016 11:06:09 +0000<br>Subject: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241<br><br>
<div dir="ltr"> Hi,<br><br>Since PJSIP Commit 5241 (<a href="https://trac.pjsip.org/repos/changeset/5241" target="_blank">https://trac.pjsip.org/repos/changeset/5241</a>) Asterisk crashes when a device registers.<br><br>The commit resolves the following:<br> <br>• Crash when endpoint has multiple worker threads and SIP TCP transport is disconnected during incoming call handling.<br>• Deprecated pjsip_dlg_create_uas(), replaced by pjsip_dlg_create_uas_and_inc_lock().<br>• Serialized transaction state notifications (of 'terminated' and 'destroyed') in case of transport error.<br> <br>This commit should resolve a previous segfault within Asterisk, however due to the deprecated method I believe this is causing an additional issue. <br> <br>Can this be easily resolved to resolve both segfaults?<br> <br>Kind regards,<br> <br>Ross<br><br><font face="Verdana"></font> <br><font face="Verdana"></font> <br><font face="Verdana"></font> <br> </div>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></div> </div>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></div></div></div> </div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank" rel="noreferrer">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank" rel="noreferrer">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br></div></div>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></div></div></div> </div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank" rel="noreferrer">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank" rel="noreferrer">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br></div></div>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev</div> </div></body>
</html>