<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">I also encountered a problem in sip recording during outbound IVR call where Freeswitch makes an outbound call to some user, and he/she answered the call to take actions following FS IVR.<br><br>for the same dialplan extension and IVR lua script, IVR runs perfect for inbound call; however, the IVR always failed to record message for outbound call although it may still play audio and detect dtmf ... For recording, a sound file is always generated on disk but it contains nothing.<br><br>For its details (dialplan and ivr lua script), please look at my post with subject [How to start IVR script via dialplan by making an outbound call].<br><br>Is it possible that sip record has some bug?<br><br>--- On <b>Tue, 8/30/11, Michael Collins <i><msc@freeswitch.org></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px;
padding-left: 5px;"><br>From: Michael Collins <msc@freeswitch.org><br>Subject: Re: [Freeswitch-users] Session ends unexpectedly during record dial plan usage<br>To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org><br>Date: Tuesday, August 30, 2011, 1:39 AM<br><br><div id="yiv2126776909"><br><br>On Monday, August 29, 2011, Anthony Minessale <<a rel="nofollow" ymailto="mailto:anthony.minessale@gmail.com" target="_blank" href="/mc/compose?to=anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>> /me punches MSC in the arm.....<br><br>Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) as well.<br>
<br>Adam,<br> If you simply want to record the call between two parties then look on the wiki for the record_session app. Look at the diff between it and the record app. They are *totally* different concepts. <br><br>Let us know if you have any other questions.<br>
<br>-MC<br>><br>> On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <<a rel="nofollow" ymailto="mailto:msc@freeswitch.org" target="_blank" href="/mc/compose?to=msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<br>>> Go ahead and get a console debug log on this along with a SIP trace. Drop it<br>
>> in pastebin. Hopefully it contains some clues as to what is happening.<br>>> -MC<br>>><br>>> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <<a rel="nofollow" ymailto="mailto:adam.kelloway@newpace.ca" target="_blank" href="/mc/compose?to=adam.kelloway@newpace.ca">adam.kelloway@newpace.ca</a>><br>
>> wrote:<br>>>><br>>>> Hi there,<br>>>><br>>>> I have a freeswitch installation that I can make sip calls to to listen<br>>>> to IVR menus. The sessions last as long as either side does not hang up.<br>
>>> The exception to this is when I use the 'record' dial plan tool. The sip<br>>>> session ends unexpectedly after about 32+ seconds into the recording.<br>>>> This happens every time I use the record tool. Note that I have set the<br>
>>> maximum message length to 120 seconds, so this shouldn't be coming into<br>>>> play here (and shouldn't affect the session anyway).<br>>>><br>>>> Has anyone ever experienced this, and do you have any suggestions as to<br>
>>> what might be the cause?<br>>>><br>>>> Note that there is no NAT involved here. There are also no Expires or<br>>>> Session-Expires header(s) in the sip INVITE or response that would<br>
>>> affect the length of the session. Indeed, the same type of session can<br>>>> continue indefinitely until about 32+ seconds after I invoke the record<br>>>> dial plan tool.<br>>>><br>
>>> Thanks,<br>>>><br>>>> Adam<br>>>><br>>>><br>>>><br>>>> FreeSWITCH-users mailing list<br>>>> <a rel="nofollow" ymailto="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank" href="/mc/compose?to=FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>> <a rel="nofollow" target="_blank" href="http://www.freeswitch.org">http://www.freeswitch.org</a><br>>><br>>><br>><br>><br>><br>> --<br>> Anthony Minessale II<br>><br>> FreeSWITCH <a rel="nofollow" target="_blank" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>
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