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Thomas Peters

2019-Jan-14 18:42 UTC

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

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.

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

2019-Jan-14 20:28 UTC

head link

[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
> -------------- next part --------------
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Thomas Peters

2019-Jan-14 21:34 UTC

head link

[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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