as a suggestion, please play a little with the next parameters in
zapata.conf read the docs in voip-info about these parameters an may
me you will be able to fix your problem.
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=-4
immediate=yes
busydetect=yes
Hello,
I am following up on a previous mail of the same subject at
http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html
In a nutshell I have connected my asterisk behind a Siemens HICOM 118E
for a small call center application. The external PSTN calls will land
in HICOM 118E
Hi,
I am setting up a test call center using *. I am using one Zap channel
(Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip
phones (SjPhone) for call agents. I have setup queues and agents. While
testing I found that if the agent presses * key in soft phone while
attending
