[Freeswitch-users] IVR Bridged Call Dropping after 2 rings

[John Carpenter]

Thanks for all the suggestions, will not be able to test until tomorrow,
will post results then

regards, John

On Tue, 2011-01-25 at 17:26 +0100, David Ponzone wrote:

> Send an audio ringback to them.
> 
> 
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
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> Le 25/01/2011 à 16:22, John Carpenter a écrit :
> 
> 
> 
> > Hi, I have now traced the problem down to the SIP tunk provider
> > having a timeout of 10 seconds. If they receive no signalling or RTP
> > for 10 seconds they drop the call. If I had known this earlier I
> > would not have signed up with them but its too late now.
> > So the question is how do I get FS to send RTP back to SIP trunk
> > when a call is being bridged, it currently dies if extension not
> > answered in 10 seconds. Have tried proxy_media and bypass_media
> > without any success. My extensions are remote and using NAT.
> > 
> > regards, John
> > 
> > On Mon, 2011-01-24 at 12:28 -0800, Michael Collins wrote:
> > 
> > > Can you pastebin a debug log with a siptrace? Also, pastebin your
> > > dialplan. I think we can help with this but I want to see what
> > > you're doing before I suggest anything.
> > > 
> > > 
> > > -MC
> > > 
> > > On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter
> > > <john at 247-talk.co.uk> wrote:
> > > 
> > >         Hi, I am trying to setup a very simple IVR using LUA. Call
> > >         arrives from a DID SIP trunk and is answered and message
> > >         is played ok, after a particular digit is pressed it
> > >         bridges the call to an extension which is remotely
> > >         connected. It works but after 2 rings the call to the
> > >         extension is dropped with a SIP message "BYE" from DID
> > >         provider. If I just route the call directly to the
> > >         extension (no IVR) it works fine. It seems like the DID
> > >         hangs up when the call is bridged to the extension. Have
> > >         tried same thing using the XML IVR Engine and get exactly
> > >         the same result. The IVR script is below
> > >         
> > >         pathsep = '/'
> > >         session:setAutoHangup(false);
> > >         session:answer()
> > >         prompt = "ivr" .. pathsep .. "247talk.wav"
> > >         invalid = "ivr" .. pathsep ..
> > >         "ivr-that_was_an_invalid_entry.wav"
> > >         freeswitch.consoleLog("INFO", "Prompt file is '" ..
> > >         prompt .. "'\n")
> > >         continue = true
> > >         
> > >         while( session:ready() == true and continue == true) do
> > >                 digits = session:playAndGetDigits(1,1,3,7000,"#",
> > >         prompt, invalid, "\\d+")
> > >                 if (digits == "1") then
> > >                         continue = false
> > >         
> > >         session:execute("bridge","sofia/external/2476%
> > >         91.xxx.xx.xx")
> > >                 end
> > >                 if (digits == "2") then
> > >         
> > >         session:execute("bridge","sofia/external/2475%
> > >         91.xxx.xx.xx")
> > >                 end
> > >                 if (digits == "3") then
> > >                         continue = false
> > >         
> > >         session:execute("bridge","sofia/external/2475%
> > >         91.xxx.xx.xx")
> > >                 end
> > >         end
> > >         
> > >         session:hangup()
> > >         
> > >         Any help with this greatly appreciated it is driving me
> > >         nuts.
> > >         
> > >         regards, John Carpenter 
> > >         
> > >         _______________________________________________
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> > > 
> > > 
> > > 
> > > 
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