[Freeswitch-users] Calls getting queued?

[Anthony Minessale]

try setting the timeout in curl

conf/autoload_configs/xml_curl.conf.xml:
<param name="timeout" value="5"/>

On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Michael,
>
> Thanks for your help. Yes, if I restart FS things go back to normal
> for a while and then the same thing starts happening again.
>
> The weird thing is, it started only 2 days ago, and happened only once
> or twice. Before that I had no trouble, and I only made 1 change,
> which I reverted, but it wasn't that. Today it's happening all the
> time, if I restart FS things will work for maybe an hour and then it
> will start doing the same thing.
>
> I'm guessing it might be something external to FS, like curl calls not
> finishing properly because of the url they are requesting or something
> like that.
>
> What kind of info should I collect? I don't think it has to do with
> sofia or any sip-related problems. I'm also using the default
> dialplan, no changes at all, I'm doing everything through JS, well and
> one really small lua script.
>
> This is the main JS file:
> It originates 2 calls and bridges them.
>
> - http://pastebin.freeswitch.org/11706
>
>
> This is another JS script which gets called when each call is hanged up:
> It gets some info and then requests a url using curl to update call
> status on an external db.
>
> - http://pastebin.freeswitch.org/11707
>
>
> This lua script calls a ruby script to do some other stuff when a call
> is answered:
>
> - http://pastebin.freeswitch.org/11708
>
>
> Thanks!
>
>
> Nico
>
>
>
> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
> > wrote:
> >>
> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
> >> to generate calls and bridge them together. Usually everything works
> >> just fine, but them at some point it's like if FS choked, calls for
> >> the first leg of the bridges are apparently made, but the second leg
> >> is never called. The call is not hanged up for several minutes and the
> >> system keeps opening new channels but never connecting a call.
> >>
> >> For example, right now, doing 'show channels' on the console, I get a
> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
> >> behavior, when everything's working fine, is to get twice as many
> >> channels as there are active calls and no channels at all when there
> >> are no calls, unless they haven't been bridged yet.
> >>
> >> The originate string is something like this:
> >>
> >> var stUsRing = "%(2000,4000,440,480)";
> >> var timeout = 45;
> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
> >> l1,execute_on_answer=lua answered.lua 1
> >>
> >>
> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
> >>
> >> Where diasltr1 has the phonenumber and and gateway info. The
> >> callback.js has a curl request to update some call info on an external
> >> database and answered.lua calls a ruby script through the os.execute()
> >> function (I know, I should be doing all this through the event socket,
> >> I was doing that but had trouble and had to come up with a quick
> >> solution).
> >>
> >> The system is not loaded at all, at least not for what I think and
> >> read that FS can handle. We are having at most 10 concurrent calls (20
> >> channels), with maybe 5 to 10 calls per minute.
> >>
> >> What worries me is not only that I don't know where the problem is,
> >> but that I have no clue how to debug it or send you guys more
> >> "lowlevel" and detailed information to give you an insight about
> >> what's going on. Any help would be greatly appreciated!
> >>
> >> Thanks!
> >>
> >> Nico
> >>
> > First off you'll want to get familiar with the resources mentioned here:
> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
> >
> > It has good tips on how to collect and report information.
> >
> > Second, I recommend that you pastebin your relevant portion of the
> dialplan
> > and the whole javascript program that you are using so that others can
> take
> > a look.
> >
> > Last thing: if you restart FreeSWITCH does everything work fine for a
> while
> > but then eventually it breaks down and exhibits the behavior that you are
> > reporting?
> >
> > -MC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100107/0f9d1ca0/attachment-0002.html