How to set V611 ATA as analog trunking on Asterisk Platform?

VOIP-->PSTN direct dialing，PSTN-->VOIP second time dialing？
1. VOIP-->PSTN，caller is VoIP terminal, dial PSTN number directly：
　a) Users can set reasonable number dialing rule on IPPBX or Softswitch system(Asterisk, latest version: 1.4.21), part of calling to certain PSTN area will be routing to the FXO of V611 ATA.(IP:5062)
b) Configure V611as shown in the image below:
Outing call to PSTN Decide whether to automatically distill the callee’s number in INVITE info to originate call from PSTN
Send DTMF to PSTN Decide originate call from PSTN firstly, then respond with a “SIP—200OK”
Send DTMF Select In-Audio in common situation, DTMF signal will be passed through directly

[img]http://www.vonets.com/img/pstn1.bmp[/img]

C) Signal attenuation is a little serious in special areas, this is possible to lead to dialing error because of DTMF transmission error, users can increase the Send Volume properly, as shown in the image below:
Redial DTMF ACG If set as “Yes”, it will automatically adjust the DTMF signal gain which is sent to PSTN line, to make sure number dialing directly.

[img]http://www.vonets.com/img/pstn2.bmp[/img]

2. PSTN-->VOIP, call any number by using second time dialing：
　 a) Setup a call center function on IPPBX or Softswitch system(Asterisk, latest version: 1.4.21), users can call any number by using second time dialing.
　 b) Configure V611 as shown in the image below(assume call center number is 8000)
The FXO allows to connect immediately express FXS ring or not when there is a PSTN incoming call.
　　 If the value of PSTN-CID to “sip” is yes in the image above, FXS will ring two times at least.

[img]http://www.vonets.com/img/pstn3.bmp[/img]

C) After step a) and b) are finished, users can use mobile phone to call the PSTN line number which is connected to FXO of V611, this call will be sent into call center automatically, users can dial the destination number according to the voice instruction of call center.


3. While V611 is set as analog trunking, the most important parameter setting is Busy Tone parameter, if users set inadequately, it may lead to the FXO disconnection failure, the FXO is always taken, can not make next call. Please refer to http://www.vonets.com/service/index_2.asp for more details about the Busy Tone parameter.