Hi,<br><br>My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP). My server is<br>connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi I get<br>DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start counting. <br>
<br>Is it normal behavior ? Can I change that ?<br><br>So channel gets in ANSWERED state and billsec starts as soon as line starts<br>to ring even if no one really pick up ordinary phone and costumer did not talk to anyone. <br>
That leads to problem that costumers will be billed even if they did not make a real <br>conversation. <br><br>How can I avoid that behavior and set asterisk to start counting billsecs after<br>someone really pick up the phone on the other side ?<br>
<br>How can I distinguish real (talking to) call from just ring (no real answer call) <br>when both are in state ANSWERED ?<br><br>I tried with timeout 20 in Dial command but since channel is "answered" when it<br>
starts to ring timeout is not doing what I want.<br><br>Here is my Dial command:<br>exten => _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:60000:20000)hH)<br><br>It works very good in case ordinary phone calls sip (for incoming calls from PSTN)<br>
because I need to click answer on xlite to move call in state ANSWERED so if I don't<br>click it is not answered and timeout works. <br><br>Can you help me with that ?<br><br>Thanks,<br>Uros<br><br><br>-- <br>Use Free Software <a href="http://www.fsf.org/">http://www.fsf.org/</a><br>
-----------------------------------------------<br>Four essential software freedoms:<br>1) To study source code<br>2) To copy program<br>3) To modify source code<br>4) To redistribute modified program under condition that new user has all 4 freedoms.<br>
Richard M. Stallman<br>