Hi, My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP). My server is connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi I get DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start counting. Is it normal behavior ? Can I change that ? So channel gets in ANSWERED state and billsec starts as soon as line starts to ring even if no one really pick up ordinary phone and costumer did not talk to anyone. That leads to problem that costumers will be billed even if they did not make a real conversation. How can I avoid that behavior and set asterisk to start counting billsecs after someone really pick up the phone on the other side ? How can I distinguish real (talking to) call from just ring (no real answer call) when both are in state ANSWERED ? I tried with timeout 20 in Dial command but since channel is "answered" when it starts to ring timeout is not doing what I want. Here is my Dial command: exten => _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:60000:20000)hH) It works very good in case ordinary phone calls sip (for incoming calls from PSTN) because I need to click answer on xlite to move call in state ANSWERED so if I don't click it is not answered and timeout works. Can you help me with that ? Thanks, Uros -- Use Free Software http://www.fsf.org/ ----------------------------------------------- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100131/20f06d34/attachment.htm
Uros Djokic wrote:> Hi, > > My costumers are logged in on my Asterisk PBX through XLite Softphone > (SIP). My server is > connected to PSTN. Problem is when SIP phone calls ordinary phone via > dahdi I get > DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is > start counting. > > Is it normal behavior ? Can I change that ? > > So channel gets in ANSWERED state and billsec starts as soon as line > starts > to ring even if no one really pick up ordinary phone and costumer did > not talk to anyone. > That leads to problem that costumers will be billed even if they did > not make a real > conversation. > > How can I avoid that behavior and set asterisk to start counting > billsecs after > someone really pick up the phone on the other side ? > > How can I distinguish real (talking to) call from just ring (no real > answer call) > when both are in state ANSWERED ? > > I tried with timeout 20 in Dial command but since channel is > "answered" when it > starts to ring timeout is not doing what I want. > > Here is my Dial command: > exten => _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:60000:20000)hH) > > It works very good in case ordinary phone calls sip (for incoming > calls from PSTN) > because I need to click answer on xlite to move call in state ANSWERED > so if I don't > click it is not answered and timeout works. > > Can you help me with that ? > > Thanks, > Uros > > > -- > Use Free Software http://www.fsf.org/ > ----------------------------------------------- > Four essential software freedoms: > 1) To study source code > 2) To copy program > 3) To modify source code > 4) To redistribute modified program under condition that new user has > all 4 freedoms. > Richard M. StallmanIt entirely depends on the technology used to interface to the PSTN. You have not specified what technology/hardware you are using to connect to the PSTN. For instance if you are using POTS(plain old telephone service - analog copper fed lines), you do not get answer supervision back from the telco. Lyle Giese LCR Computer Services, Inc.
It entirely depends on the technology used to interface to the PSTN.> You have not specified what technology/hardware you are using to connect > to the PSTN. > > For instance if you are using POTS(plain old telephone service - analog > copper fed lines), you do not get answer supervision back from the telco.-- > >I am using tdm400 card with one fxo port. I am using analog line so I guess it's POTS "analog cooper fed" line. So it is impossible to distinguish ringing from talking and billsec must start when ringing begin due missing answer supervision from telco ? Thanks for reply Use Free Software http://www.fsf.org/ ----------------------------------------------- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100201/17b62211/attachment-0001.htm