On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters <tpet...@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@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 | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@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