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> On 15/01/2019, at 10:34 AM, Thomas Peters <[email protected]> wrote: > > Duncan: > > 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. > > I wonder how I can change the timing source. In one version (and I can’t recall which) asterisk moved to an internal timing system, to avoid the hardware need. There should be quite a lot of discussion of it in the archives or perhaps voipinfo 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 Cheers Duncan > > Thomas M. Peters | Sr. Systems Administrator | [email protected] > Desk: 414.343.1720 | Helpdesk: x3400 or [email protected] > Milwaukee County Transit System > > 1942 N 17th Street | Milwaukee, WI 53205 > Check us out on Facebook & Twitter > > From: asterisk-users <[email protected]> On Behalf Of > Duncan > Sent: Monday, January 14, 2019 2:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail > > > > On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters <[email protected]> wrote: > > 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. > > 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 > > > > Now, random extensions ring once and go straight to voicemail. > > 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. > > 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 > app_dial.c – SIP/1234-00001111 is ringing > Then eventually > app_dial.c: -- Nobody picked up in 30000 ms > > But according to the timestamps, the time from the one event to the other is > ten seconds! > > Here’s another example- call starts: > [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 > . . . > [2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-00001266 is > ringing > . . . > [2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in > 20000 ms > So again, the elapsed time is nowhere near 20 seconds. > > Another: This time the ring time has been set to 30 seconds (and I still get > only 2 rings) > [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 > . . . > [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 > . . . > [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 > . . . > [2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- > SIP/3327-00001305 is ringing > . . . > [2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- > Nobody picked up in 30000 ms > > So, after 9 seconds, it says “Nobody picked up after 30000 ms”??? > > Is this some weirdness of Oracle VMs? Anybody have any advice for me? > > > Additional information: > FreePBX version 2.9.0.7 > PBX in a Flash Version 1.2 Daemon Status > ******************************************************************** > * Asterisk * ONLINE * Dahdi * ONLINE * MySQL * ONLINE * > * SSH * ONLINE * Apache * ONLINE * Iptables * OFFLINE * > * Fail2ban * OFFLINE * IP Connect* ONLINE * Ip6tables * OFFLINE * > * BlueTooth * ONLINE * Hidd * ONLINE * NTPD * ONLINE * > * Sendmail * ONLINE * Samba * OFFLINE * Webmin * LOADING * > * Ethernet0 * ONLINE * Ethernet1 * ONLINE * Wlan0 * N/A * > ******************************************************************** > * Running Asterisk Version : Asterisk 1.8.7.0 > * Asterisk Source Version : 1.8.7.0 > * Dahdi Source Version : 2.5.0.1+2.5.0.1 > * Libpri Source Version : 1.4.12 > * Addons Source Version : 1.4.7 > ******************************************************************** > Voipserver on 10.10.141.251 - eth0 > Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: > 2.6.18-92.1.6.el5 > > > > Thomas M. Peters | Sr. Systems Administrator | [email protected] > Desk: 414.343.1720 | Helpdesk: x3400 or [email protected] > Milwaukee County Transit System > > 1942 N 17th Street | Milwaukee, WI 53205 > Check us out on Facebook & Twitter > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
