Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Mark D. Anderson <mda@discerning.com> To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, March 20, 2007 11:55:42 PM Subject: [Freeswitch-users] two problems with mod_conference I'm currently running head (4687) but these problems have been unchanged since I've started playing with mod_conference about 2 months ago. I have tried a cross-product of two ITSPs (vitelity and teliax) either landline caller or cell caller mod_conference "native", or a javascript based on confroom.js I have found some differences in behavior in these 8 cases. 1. Missing 1 second of initial audio (callers don't hear the beginning of the "please enter your pin" message). It doesn't matter whether I move the session.answer() earlier. This happens except for {teliax, landline, *} You can avoid this by adding a sleep action to your dialplan first answer the call from the dialplan and then use sleep to run off 1.5 lead-in seconds to avoid missing some of the audio <action application="answer"/> <action application="sleep" data="1500"/> <action application="conference" data="???"/> 2. Doubling (and tripling) of incoming dtmf digits. It is quite hard to get it to accept a pin, because it keeps thinking I've entered a digit twice, when I haven't. As far as I can see, there is no user configuration to tune dtmf configuration (like min/max length of a tone, or echo cancellation). I also don't know if there is a way to force freeswitch/sofia to use only a particular dtmf system. Note that the problem seems worse when running javascript confroom.js, but it happens both ways. This happens with both ITSPs. I see it except for {*, landline, native mod_conference.c}. Like Mike said, we are only getting the DTMF as rfc 2833 and it has been tested to parse the packets with a duration of as little as 1ms up to several seconds with accuracy. As a receiver of rfc2833 there is really nothing to tune but I know all too well from to much time using rtp that a disagreement in the protocol could cause some issues so I recommend you take Mike up on his offer to investigate it. Any suggestion for causes, fixes, ways to diagnose further, etc., are welcome. -mda _______________________________________________ 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 ____________________________________________________________________________________ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front