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Thanks for all the suggestions, will not be able to test until tomorrow, will post results then<BR>
<BR>
regards, John<BR>
<BR>
On Tue, 2011-01-25 at 17:26 +0100, David Ponzone wrote:<BR>
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Send an audio ringback to them.
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<FONT COLOR="#1c00ff">David Ponzone </FONT><FONT COLOR="#000000">Direction Technique</FONT>
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email: <A HREF="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</A>
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tel: 01 74 03 18 97
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gsm: 06 66 98 76 34
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<FONT COLOR="#1c00ff">Service Client </FONT><FONT COLOR="#ff0000">IP</FONT><FONT COLOR="#1c00ff">eva</FONT>
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<FONT COLOR="#1c00ff">tel: 0811 46 26 26</FONT>
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<FONT COLOR="#1c00ff"><A HREF="BLOCKED::http://www.ipeva.fr/">www.ipeva.fr</A> - <A HREF="BLOCKED::http://www.ipeva-studio.com/">www.ipeva-studio.com</A></FONT>
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<I><FONT COLOR="#1c00ff">Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </FONT></I><I><FONT COLOR="#1c00ff"><B>IPeva</B></FONT></I><I><FONT COLOR="#1c00ff"> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</FONT></I>
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Le 25/01/2011 à 16:22, John Carpenter a écrit :
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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>
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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.<BR>
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-MC<BR>
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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:<BR>
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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, "<A HREF="smb://d+">\\d+</A>")<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 <BR>
<BR>
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