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[stevecrozz at gmail.com]

We had our first big issues with our freeswitch system today. During at least 2 conferences, audio became jittery and there were three occasions where everyone was dropped from a conference. Even so, conference recording was not interrupted, and the freeswitch debug log doesn't show anything unusual.

Our hardware monitoring software doesn't show any kind of unusual resources usage, and our web host claims there were no outages during the time when we were experiencing problems.

We're currently running revision 12259.

How should I proceed in diagnosing this issue?

--Stephen

[jason at jasonjgw.net]

Stephen Crosby <stevecrozz@gmail.com> wrote:

[stevecrozz at gmail.com]

Network problem is what I'm still thinking. Take a look at this log snippet:
http://pastebin.freeswitch.org/8813

I'm CALLER_A, and you can see me calling back in on line 8 after I got dropped. But that was only seconds prior. There really seems to be nothing in the log at the time the calls were dropped.

--Stephen

On Mon, May 4, 2009 at 10:27 PM, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:

[stevecrozz at gmail.com]

To answer the other question, I did not try pinging the host while the problems were occurring, but I was able to call back in with no problem only a few seconds after I got dropped (less than 20), so if it is a network issue it would have to have been very brief.

--Stephen

On Mon, May 4, 2009 at 10:52 PM, Stephen Crosby <stevecrozz@gmail.com (stevecrozz@gmail.com)> wrote:

[stevecrozz at gmail.com]

Sorry for the extra messages, but I've just discovered something that probably helps pinpoint the problem:
http://pastebin.freeswitch.org/8814

It seems that the callers were disconnected, but freeswitch had to wait a timeout period before it actually hangs up which looks like about 5 minutes. So I was disconnected from a conference, then I called back in, then freeswitch later hung up on the first call. This seems very much like a network problem.

What can I do to fix it?

--Stephen

On Mon, May 4, 2009 at 10:55 PM, Stephen Crosby <stevecrozz@gmail.com (stevecrozz@gmail.com)> wrote:

[jason at jasonjgw.net]

Stephen Crosby <stevecrozz@gmail.com> wrote:

[jason at jasonjgw.net]

Stephen Crosby <stevecrozz@gmail.com> wrote:

[stevecrozz at gmail.com]

I was on this conference and there were 4 of us that got dropped at the same time. By the way, these were all regular PSTN lines calling through our SIP provider and being routed to our freeswitch instance which is being hosted on a VPS.

When you say a device on my end terminated the call, do you mean our home telephones which we used to dial in or the freeswitch box?

--Stephen

On Mon, May 4, 2009 at 11:22 PM, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:

[jason at jasonjgw.net]

Stephen Crosby <stevecrozz@gmail.com> wrote:

[brian at freeswitch.org]

First off you're not on SVN trunk secondly Are you executing the conference app inside your js file? If so then there could be the problem! You have also forgotten to include anything about Distro, OS, CPU and Memory.

/b

On May 5, 2009, at 12:12 AM, Stephen Crosby wrote:

[stevecrozz at gmail.com]

I know I'm not on svn trunk, but this is a production server and it's just not feasible to update it constantly. I can update it though if you think I need to. I am routing callers to the conference app with javascript like this:
session.execute("conference", xyz);
Can you tell me more about the problems I could have?

The machine running freeswitch has 1024MB memory and I'm not sure about the CPU since its a VPS.

--Stephen

On Tue, May 5, 2009 at 3:49 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[stevecrozz at gmail.com]

Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron).

--Stephen

On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby <stevecrozz@gmail.com (stevecrozz@gmail.com)> wrote:

[anthony.minessale at g...]

You should rule out the network problems first, which sound more likely.

you can reduce the overuse of JS if you transfer the call to a regular extension with a dynamic regex.

session.execute("transfer", "conf-xyz");

then make a regex in your xml dialplan to pick up ^conf-(.*) and execute conference $1


On Tue, May 5, 2009 at 9:48 AM, Stephen Crosby <stevecrozz@gmail.com (stevecrozz@gmail.com)> wrote:

[mfedyk at mikefedyk.com]

Why is initiating a conference from a dial plan entry better than
directly from an internal script? Is there a page that details what
is and isn't script abuse?

On 5/5/09, Anthony Minessale <anthony.minessale@gmail.com> wrote:

[msc at freeswitch.org]

Javascript has it's own issues. Tony wrote FreeSWITCH, including the dialplan parser. The less you can use JS, especially in a high-volume environment, the better. The happy medium is to use Lua. It is small, lightweight, easy, and designed to be embedded. Tony has had 3000 Lua session up simultaneously with no issues. I guarantee you that JS would have a meltdown at a fraction of that volume...

-MC

On Thu, May 14, 2009 at 9:32 AM, Mike Fedyk <mfedyk@mikefedyk.com (mfedyk@mikefedyk.com)> wrote: