<div>Hello all,</div>
<div>Hoping someone can help me with an issue...I have i .call file which calls out on a SIP channel and connects to an extension which dials another SIP channel. (both via voip providers) - both to PSTN.</div>
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<div>Problem is, hanging up the POTS phone doesn't release the channel (either one - hanging up the calling channel or the destination doesn't do it).</div>
<div> </div>
<div>Using IAX instead of SIP works better; it releases the voip channels but not the POTS channels (i.e. the POTS phones don't immediately go back to a dial tone or fast busy).</div>
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<div>I'm using asterisk 1.4</div>
<div>here are the relevant bits:</div>
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<div>the extension:</div>
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<div>exten=> 157,1,Answer<br>exten=> 157,2,Dial(<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:IAX2/myusername@voipjet/PSTNnumberToCall2" target="_blank">IAX2/myusername@voipjet/PSTNnumberToCall2
</a>, 60)<br>exten=> 157,3,Hangup</div>
<div> </div>
<div>the .call file:</div>
<div>
<p># Create the call on group 2 dial lines and set up<br># some re-try timers<br>#<br>Channel: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:IAX2/username@voipjet/PSTNnumberToCall1" target="_blank">
IAX2/username@voipjet/PSTNnumberToCall1</a><br>MaxRetries: 2<br>RetryTime: 60 <br>WaitTime: 30<br>Extension: 157<br>Priority: 1<br></p>
<p>if anyone can shed some light on this i'd be eternally grateful, first of all, why if i glue 2 SIP channels together hanging up the POTS phone doesn't release the SIP channels, and second why if i glue two IAX channels together it doesnt release the POTS lines.
</p>
<p>thanks,</p>
<p> yair</p>
<p> </p></div>