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<DIV><FONT size=2>what's happen if you send comfort_noise ?</FONT></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=john@247-talk.co.uk href="mailto:john@247-talk.co.uk">John
Carpenter</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, January 25, 2011 10:22
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] IVR
Bridged Call Dropping after 2 rings</DIV>
<DIV><BR></DIV>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.<BR>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.<BR><BR>regards, John<BR><BR>On Mon, 2011-01-24 at 12:28 -0800, Michael
Collins wrote:<BR>
<BLOCKQUOTE TYPE="CITE">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. </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><BR><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">-MC<BR><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">On Fri, Jan 21, 2011 at 5:57 PM, John Carpenter
<<A href="mailto:john@247-talk.co.uk">john@247-talk.co.uk</A>> wrote:
</BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE>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<BR><BR>pathsep =
'/'<BR>session:setAutoHangup(false);<BR>session:answer()<BR>prompt = "ivr"
.. pathsep .. "247talk.wav"<BR>invalid = "ivr" .. pathsep ..
"ivr-that_was_an_invalid_entry.wav"<BR>freeswitch.consoleLog("INFO",
"Prompt file is '" .. prompt .. "'\n")<BR>continue = true<BR><BR>while(
session:ready() == true and continue == true)
do<BR> digits =
session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid,
"\\d+")<BR> if (digits == "1")
then<BR>
continue =
false<BR>
session:execute("bridge","sofia/external/2476%91.xxx.xx.xx")<BR>
end<BR> if (digits == "2")
then<BR>
session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")<BR>
end<BR> if (digits == "3")
then<BR>
continue =
false<BR>
session:execute("bridge","sofia/external/2475%91.xxx.xx.xx")<BR>
end<BR>end<BR><BR>session:hangup()<BR><BR>Any help with this greatly
appreciated it is driving me nuts.<BR><BR>regards, John Carpenter
</BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
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