<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><br><br><div id="AppleMailSignature" dir="ltr">Sent from my iPad</div><div dir="ltr"><br>On 15/01/2019, at 10:34 AM, Thomas Peters <<a href="mailto:TPeters@mcts.org">TPeters@mcts.org</a>> wrote:<br><br></div><blockquote type="cite"><div dir="ltr">
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<p class="MsoNormal">Duncan: <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">You may have it right—I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one.
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I wonder how I can change the timing source. </p></div></div></blockquote><div><br></div><div>In one version (and I can’t recall which) asterisk moved to an internal timing system, to avoid the hardware need.</div><div><br></div><div>There should be quite a lot of discussion of it in the archives or perhaps voipinfo</div><div><br></div><div>I don’t know if you can slow the VM processor speed? I am guessing it is counting something much faster than it used to</div><div><br></div><div>Cheers Duncan</div><div><br></div><div><br></div><br><blockquote type="cite"><div dir="ltr"><div class="WordSection1"><p class="MsoNormal"><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:#006FC9">Thomas M. Peters | Sr. Systems Administrator |
<a href="mailto:tpeters@mcts.org"><span style="color:blue">tpeters@mcts.org</span></a>
<br>
Desk: 414.343.1720 | Helpdesk: x3400 or <a href="mailto:helpdesk@mcts.org"><span style="color:blue">helpdesk@mcts.org</span></a></span><span style="font-size:9.0pt;color:black"><o:p></o:p></span></p>
<p class="MsoNormal" style="background:white"><u><span style="font-size:9.0pt;color:#006FC9"><a href="http://www.ridemcts.com/"><span style="color:#006FC9">Milwaukee County Transit System
</span></a></span></u><u><span style="font-size:9.0pt;color:blue"><o:p></o:p></span></u></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:black"><o:p> </o:p></span></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:#006FC9">1942 N 17th Street | Milwaukee, WI 53205</span><span style="font-size:9.0pt;color:black"><o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>From:</b> asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>>
<b>On Behalf Of </b>Duncan<br>
<b>Sent:</b> Monday, January 14, 2019 2:29 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
<b>Subject:</b> Re: [asterisk-users] Various extensions ring once and go to voicemail<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters <<a href="mailto:TPeters@mcts.org">TPeters@mcts.org</a>> wrote:<br>
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<p class="MsoNormal">We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it.
<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:12.0pt">Thats a while back, I think it tended to use zaptel or dahdi hardware as a timer, you may need to find a timing source as perhaps the clock in the VM is just going too fast<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal">Now, random extensions ring once and go straight to voicemail.
<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">I went to one of the affected extensions and changed the ring time from the default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. Other extensions ring once or twice. After some time has gone by since this was first
reported, all phones in my random sample ring only twice.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">As I trace a call to that extension through the log, I notice it setting the ring timer properly (I think) and then I see
<o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">app_dial.c – SIP/1234-00001111 is ringing<o:p></o:p></p>
<p class="MsoNormal">Then eventually <o:p></o:p></p>
<p class="MsoNormal"> app_dial.c: -- Nobody picked up in 30000 ms<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">But according to the timestamps, the time from the one event to the other is ten seconds!<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Here’s another example- call starts: <o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing [3327@cc-long-distance:1] ExecIf("SIP/4704-00001265", "0?Set(__RINGTIMER=0)") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">. . . <o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-00001266 is ringing<o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">. . . <o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in 20000 ms<o:p></o:p></p>
<p class="MsoNormal">So again, the elapsed time is nowhere near 20 seconds. <o:p>
</o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Another: This time the ring time has been set to 30 seconds (and I still get only 2 rings)<o:p></o:p></p>
<p class="MsoNormal" style="text-indent:.5in">[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing [3327@cc-long-distance:1] ExecIf("SIP/4704-00001304", "1?Set(__RINGTIMER=30)") in new stack<o:p></o:p></p>
<p class="MsoNormal"> . . . <o:p></o:p></p>
<p class="MsoNormal"> [2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/4704-00001304", "RT=30") in new stack<o:p></o:p></p>
<p class="MsoNormal"> . . . <o:p></o:p></p>
<p class="MsoNormal"> [2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing [s@macro-dial-one:30] Set("SIP/4704-00001304", "D_OPTIONS=trWw") in new stack<o:p></o:p></p>
<p class="MsoNormal"> . . .<o:p></o:p></p>
<p class="MsoNormal"> [2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- SIP/3327-00001305 is ringing<o:p></o:p></p>
<p class="MsoNormal"> . . .<o:p></o:p></p>
<p class="MsoNormal"> [2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- Nobody picked up in 30000 ms<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">So, after 9 seconds, it says “Nobody picked up after 30000 ms”???<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Is this some weirdness of Oracle VMs? Anybody have any advice for me?
<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Additional information: <o:p></o:p></p>
<p class="MsoNormal">FreePBX version 2.9.0.7<o:p></o:p></p>
<p class="MsoNormal"> PBX in a Flash Version 1.2 Daemon Status<o:p></o:p></p>
<p class="MsoNormal">********************************************************************<o:p></o:p></p>
<p class="MsoNormal">* Asterisk * ONLINE * Dahdi * ONLINE * MySQL * ONLINE *<o:p></o:p></p>
<p class="MsoNormal">* SSH * ONLINE * Apache * ONLINE * Iptables * OFFLINE *<o:p></o:p></p>
<p class="MsoNormal">* Fail2ban * OFFLINE * IP Connect* ONLINE * Ip6tables * OFFLINE *<o:p></o:p></p>
<p class="MsoNormal">* BlueTooth * ONLINE * Hidd * ONLINE * NTPD * ONLINE *<o:p></o:p></p>
<p class="MsoNormal">* Sendmail * ONLINE * Samba * OFFLINE * Webmin * LOADING *<o:p></o:p></p>
<p class="MsoNormal">* Ethernet0 * ONLINE * Ethernet1 * ONLINE * Wlan0 * N/A *<o:p></o:p></p>
<p class="MsoNormal">********************************************************************<o:p></o:p></p>
<p class="MsoNormal">* Running Asterisk Version : Asterisk 1.8.7.0<o:p></o:p></p>
<p class="MsoNormal">* Asterisk Source Version : 1.8.7.0<o:p></o:p></p>
<p class="MsoNormal">* Dahdi Source Version : 2.5.0.1+2.5.0.1<o:p></o:p></p>
<p class="MsoNormal">* Libpri Source Version : 1.4.12<o:p></o:p></p>
<p class="MsoNormal">* Addons Source Version : 1.4.7<o:p></o:p></p>
<p class="MsoNormal">********************************************************************<o:p></o:p></p>
<p class="MsoNormal">Voipserver on 10.10.141.251 - eth0<o:p></o:p></p>
<div style="border:none;border-bottom:dotted windowtext 3.0pt;padding:0in 0in 1.0pt 0in">
<p class="MsoNormal">Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: 2.6.18-92.1.6.el5<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:#006FC9">Thomas M. Peters | Sr. Systems Administrator |
<a href="mailto:tpeters@mcts.org"><span style="color:blue">tpeters@mcts.org</span></a>
<br>
Desk: 414.343.1720 | Helpdesk: x3400 or <a href="mailto:helpdesk@mcts.org"><span style="color:blue">helpdesk@mcts.org</span></a></span><o:p></o:p></p>
<p class="MsoNormal" style="background:white"><u><span style="font-size:9.0pt;color:#006FC9"><a href="http://www.ridemcts.com/"><span style="color:#006FC9">Milwaukee County Transit System
</span></a></span></u><o:p></o:p></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:black"> </span><o:p></o:p></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:#006FC9">1942 N 17th Street | Milwaukee, WI 53205</span><o:p></o:p></p>
<p class="MsoNormal" style="background:white"><span style="font-size:9.0pt;color:#006FC9">Check us out on
<a href="https://www.facebook.com/mcts"><span style="color:blue">Facebook</span></a> &
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<p class="MsoNormal"> <o:p></o:p></p>
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</div></blockquote><blockquote type="cite"><div dir="ltr"><span>-- </span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span></span><br><span>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/">https://community.asterisk.org/</a></span><br><span></span><br><span>New to Asterisk? Start here:</span><br><span> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a></span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></div></blockquote></body></html>