2007/12/7, C F <[EMAIL PROTECTED]>: > by 3rd call do you mean over the internet? > if the answer is yes, then I wouldn't be surprised.
Oh my god! If it is over internet and you get crap quality.. you have to be surprised.. It is depends by Latency (Traffic congestion, Network congestion) and Packet loss --------------------------------------------------------------------------------- jorain, What do you mean for "quality problem" ? Different "quality" problems are generated by different parameter braking ? echo? low volume ? Cheers 2007/12/7, C F <[EMAIL PROTECTED]>: > by 3rd call do you mean over the internet? > if the answer is yes, then I wouldn't be surprised. another thing what > codec are you using? > > On 12/6/07, jorain <[EMAIL PROTECTED]> wrote: > > Hi all, > > > > We are using > > - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size > > bus 2MB cache) as the asterisk server > > - dell 400sc(Intel P4) as a SER server > > - digium isdn card, TE120P at Asterisk server > > - Bandwidth: 2Mbps/512kbps > > > > All SIP Phones are registered to SER server, and SER will route all outgoing > > calls to Asterisk server. My problem is the sound quality goes down if more > > than 3 concurent calls to PSTN. > > > > Logically i think our system and bandwidth are more than enough to handle 3 > > concurent calls, but as the 4th person use it, the sound become jerky and a > > bit delay. So how can we improve the sound quality? > > > > > > Thanks > > > > Regards, > > jorain > > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Giovanni Miano _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users