[asterisk-users] Various extensions ring once and go to voicemail

Thomas Peters TPeters at mcts.org
Mon Jan 14 15:34:17 CST 2019


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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