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

[Wesley Akio]

It's 2 in the morning and I'm really sleepy so pardon me if it does not
make a lot of sense but maybe something to do with php timeout closing the
socket and dropping the call?
Em 22/07/2013 14:33, "Michael Collins" <msc at freeswitch.org> escreveu:

> 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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>>
>> 
>> 
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>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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