[Asterisk-Users] audio delay in meetme conference using ztdummy

[Senad Jordanovic]

Davin O'Neill wrote:
> I have Asterisk running on a Linux 2.4.x box with ztdummy.  Once I
> did a modprobe on ztdummy I was able to enter into a conference room
> using my softphone clients.  I'm using SJphone and Firefly.  I have
> noticed a significant delay (1 to 3 seconds) while talking within the
> conference room.  I have tried both clients, SIP and IAX protocols
> and various codecs.  I have also tried it from different host
> machine.  They are all on the same LAN, so that shouldn't be an
> issue.  I can call a client directly with SIP or IAX and have clear,
> timely audio.  I have also done echo tests (dialing 600) through
> Asterisk and that works fine too.  The delay only occurs within the
> conference room.  I'm wondering if I just need to purchase one of the
> zaptel cards.  I would appreciate any thoughts or suggestions.       
> 
> Thanks!

try adding "q" flag to meetme app ...