Thomas Peters
2019-Jan-14 21:34 UTC
[asterisk-users] Various extensions ring once and go to voicemail
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.
Thomas M. Peters | Sr. Systems Administrator | tpeters at
mcts.org<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk
at mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>
1942 N 17th Street | Milwaukee, WI 53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter
<https://twitter.com/RideMCTS>
From: asterisk-users <asterisk-users-bounces at lists.digium.com> On
Behalf Of Duncan
Sent: Monday, January 14, 2019 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail
On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters <TPeters at
mcts.org<mailto:TPeters at mcts.org>> 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 at
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 at
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
at 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
at 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 | tpeters at
mcts.org<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk
at mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>
1942 N 17th Street | Milwaukee, WI 53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter
<https://twitter.com/RideMCTS>
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Duncan Turnbull
2019-Jan-14 22:02 UTC
[asterisk-users] Various extensions ring once and go to voicemail
Sent from my iPad> On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org> 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 | tpeters at mcts.org > Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org > Milwaukee County Transit System > > 1942 N 17th Street | Milwaukee, WI 53205 > Check us out on Facebook & Twitter > > From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Duncan > Sent: Monday, January 14, 2019 2:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail > > > > On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters <TPeters at mcts.org> 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 at 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 at 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 at 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 at 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 | tpeters at mcts.org > Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org > 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190115/eb5b44f9/attachment.html>
Carlos Chavez
2019-Jan-14 22:08 UTC
[asterisk-users] Various extensions ring once and go to voicemail
On 1/14/19 4:02 PM, Duncan Turnbull wrote:> > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> 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 > >*CLI> module show like timing Module Description Use Count Status Support Level res_timing_dahdi.so DAHDI Timing Interface 0 Running core res_timing_pthread.so pthread Timing Interface 0 Running extended res_timing_timerfd.so Timerfd Timing Interface 1 Running core 3 modules loaded This will show you what module Asterisk is using for timing. You can try doing a noload on the two you do not need. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190114/f608670e/attachment.html>