If using record, try setting this in the dialplan first; <action application="set" data="record_waste_resources=true"/> This will force FS to send RTP - which might be the cause of the problem. /Peter Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Adam Kelloway Skickat: den 30 augusti 2011 14:10 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Session ends unexpectedly during record dial plan usage This was reproduced when making a call to FS using a Linphone SIP client. The SIP client displays a message saying that the session was ended by the remote peer (FS) unexpectedly. A SIP trace shows, however, that the Linphone client ended it unexpectedly (sent the BYE). I haven't seen this when calling from other user agents. The FS logs didn't show anything out of the ordinary. I only use that client for testing anyway. If anyone has it installed, they can try out this scenario for their own curiosity and see if you see the same behavior. In any case, thanks for the reply, Adam On 3:59 PM, Michael Collins wrote: On Monday, August 29, 2011, Anthony Minessale <anthony.minessale@gmail.com<mailto:anthony.minessale@gmail.com>> wrote:
/me punches MSC in the arm.....
Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) as well. Adam, 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. Let us know if you have any other questions. -MC
On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <msc@freeswitch.org<mailto:msc@freeswitch.org>> wrote:
Go ahead and get a console debug log on this along with a SIP trace. Drop it in pastebin. Hopefully it contains some clues as to what is happening. -MC
On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway@newpace.ca<mailto:adam.kelloway@newpace.ca>> wrote:
Hi there,
I have a freeswitch installation that I can make sip calls to to listen to IVR menus. The sessions last as long as either side does not hang up. The exception to this is when I use the 'record' dial plan tool. The sip session ends unexpectedly after about 32+ seconds into the recording. This happens every time I use the record tool. Note that I have set the maximum message length to 120 seconds, so this shouldn't be coming into play here (and shouldn't affect the session anyway).
Has anyone ever experienced this, and do you have any suggestions as to what might be the cause?
Note that there is no NAT involved here. There are also no Expires or Session-Expires header(s) in the sip INVITE or response that would affect the length of the session. Indeed, the same type of session can continue indefinitely until about 32+ seconds after I invoke the record dial plan tool.
Thanks,
Adam
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