[Freeswitch-users] Calls from api originate disconnect after 5 minutes

[Michael Collins]

Hi Fraser,

I'm curious about this line (#842 from PB 21211):
2013-07-20 10:42:15.093114 [NOTICE] mod_commands.c:2960 Hangup
sofia/internal/sip:12605 at 192.168.1.43:51977 [CS_EXECUTE] [NORMAL_CLEARING]

That hangup is coming from mod_commands.c, which suggests that there is
something going on related to the originate API. For test purposes, what
happens when you manually perform the originate from fs_cli without using
the web/PHP stuff? See if the symptom occurs there or not and that may help
narrow down where to look next.

-MC



On Sat, Jul 20, 2013 at 5:15 AM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Some of our calls are getting disconnected after exactly 5 minutes. I've
> finally narrowed it down to be something to do with how the calls are
> initiated. We have 2 ways of initiating calls:
>
> 1) Direct from a softphone out to a gateway to a landline number - this
> doesn't disconnect after 5min
>
> 2) From our webapp using php and fsock to call a command like this:
>     api originate {otherVarsGetPassedThruHereFromPhp=x}user/12605@$freeswitchDomain
> 442030112233
> (I pass through about 8-10 vars. I've tried removing them all, but it
> still disconnects.)
> The call works fine and everything is normal until 5 minutes after the
> call was answered, at which point it hangs up (usually with a second or
> two, sometimes up to 10 seconds after the 5 minute mark.)
>
>
>
> A few things I've already tried and ruled out:
>
> - I've stripped out all the javascript files that we run, variables we
> set, and commands we run, so that the only part of the dialplan that gets
> executed are just:
>     <action application="bridge"
> data="sofia/gateway/flowroute/796000#442030112233"/>
>
> - Right before it disconnects there are no extra lines in the sip log
> (other than those to handle the hangup.) There's also no sip messages in
> the log immediately before it disconnects.
>
> - A year ago I set "record_waste_resources=true" and that seemed to fix it
> at the time - but it may have only fixed the calls direct from the
> softphone.
>
> - In the internal and external profiles I've set rtp-timeout-sec=3600 (it
> used to be 300, and I thought that might be the cause.)
>
> - I've done a file compare between the log outputs of each type of call
> and theres differences that I'd expect because of the different types of
> call, but otherwise they're about the same. (Differences in how the codec
> gets set, and a slightly different route through the dialplan.)
>
>
>
> A few other details about our setup
>
> - I upgrade to the latest version of freeswitch regularly, but the problem
> has been happening for months or years.
>
> - Our production server is on Amazon AWS, so there could be a NAT issue...
> but it also happens on my dev server on my local network (though that is
> behind a NAT too, so it could be the NAT between my dev server and the
> gateway... what would I do about that? And why would it happen with one
> type of call, but not the other.)
>
> - Normally, all our calls are recorded, so I'm not doing bypass-media. The
> wav file for a 5min recording is just slightly under 10mb, so I had been
> wondering if that was the limit being hit, but it still disconnects if
> recording is off.
>
> - I've tried with a different gateway and different softphone, and it
> still happens.
>
> - Here's a full log output (with sip) if you want to take a look:
>
> http://pastebin.freeswitch.com/21211
> http://pastebin.freeswitch.com/21212 (this is direct from the softphone,
> in case the comparison helps)
>
>
> I'm stumped, so any ideas or advice would be much appreciated. I have a
> pcap I can send if you want to look at one.
>
> My best guess right now is that it's a bug to do with api originate. (That
> it's setting a variable that a normal call doesn't, or vice-versa.)
>
>
> Or is there something I should change about the format of either:
>     user/12605@$freeswitchDomain
> or
>     sofia/gateway/flowroute/796000#442030112233
> (I set up the format of both of those strings about 3 years ago, so maybe
> there's a better way to do it now.)
>
> Cheers,
> Fraser
>
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
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