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<DIV><SPAN class=139160019-22032005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
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size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=139160019-22032005><FONT face=Arial
size=2>Thanks!</FONT></SPAN></DIV></BODY></HTML>