Hi regarding this,<br> <br>I receive the calls form sip then forward it to ss7 can this help you ? as there is no need to release the channel.<br><br>[from-sip]<br>exten =>_NXXXXXXXXX,1,Set(GROUP()=VOICECHAT) <br>;Set Group<br>
exten =>_NXXXXXXXXX,2,GotoIf($[${GROUP_COUNT()} > ${VOIPMAX}]?103) <br>exten =>_NXXXXXXXXX,3,Dial(dahdi/g1/${EXTEN})<br>exten =>_NXXXXXXXXX,4,Hangup<br>exten =>_NXXXXXXXXX,103,Hangup<br><br><br><br><div class="gmail_quote">
On Fri, Jun 11, 2010 at 4:05 AM, Jorge Valencia <span dir="ltr"><<a href="mailto:jvalencia@chile.com">jvalencia@chile.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div><font face="Arial" size="2"><span>Hi guys,
</span></font></div>
<div><font face="Arial" size="2"><span></span></font> </div>
<div><font face="Arial" size="2"><span>i need to work out
on something to do the following</span></font></div>
<div><font face="Arial" size="2"><span></span></font> </div>
<div><font face="Arial" size="2"><span>receive a call from
pstn to asterisk+libss7 and then send it to a SIP IVR, some calls need to be
redirected to pstn, so i dont know how to release the channel between SIP IVR
machine and Asterisk, is there a way to do it without reprogramming libss7 or
asterisk stack?</span></font></div>
<div><font face="Arial" size="2"><span></span></font> </div>
<div> </div>
<div align="left"><font face="Arial" size="2"><span>Regards</span></font></div>
<div align="left"><font face="Arial" size="2"></font> </div>
<div align="left"><font face="Arial" size="2">Jorge Valencia Gallardo</font></div>
<div align="left"><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"></font> </div></div>
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