should be quite easy to do, probably a day of scripting. The only issue, how to measure the RTP quality in real time. ________________________________ From: Victor Chukalovskiy <victor.chukalovskiy@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Friday, September 27, 2013 6:57 PM Subject: [Freeswitch-users] Help needed with audio test setup Hello Community, The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor. Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list. Here is the idea: -FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off. Thank you, Victor _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org