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

[Tim Pushor]

I have noticed the same thing, and I have a tdm400p. I think others are 
having this issue as well, and I havn't tackled it yet .. just so you 
know that buying hardware may not fix it..

Senad Jordanovic wrote:

>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 ...
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>