Mmm. I doubt that it is an audio level problem. Look at the picture that wireshark "voip calls -> player" shows for that wheezy rbt. Here it is: This graph shows that FS really plays "stutter" tone instead of normal rbt. Or am I mistaken? And wireshark analysis of that rtp stream is strange.it shows 22 _seconds_ for that rtp stream.. Here is the text summary of wireshark rtp stream analysis of the stream from FS to IPO: Max delta = 50,65 ms at packet no. 1147 Max jitter = 22556,56 ms. Mean jitter = 238,60 ms. Max skew = 355392,29 ms. Total RTP packets = 1538 (expected 1538) Lost RTP packets = 0 (0,00%) Sequence errors = 0 Duration 36,14 s (513134 ms clock drift, corresponding to 121603 Hz (+1420,03%) I think the problem is on FS site. Thanks and regards, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 6:40 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working It looks like audio level issue .. can you lower the gain on IPO ? T. On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond@nstel.ru> wrote: Sent again.. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 4:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond@nstel.ru> wrote: Tihomir, I've just sent the trace to your gmail address. didn't get anything... Nikolay. _______________________________________________ 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
