<div><font face="Arial" size="2"> <div><font face="Arial" size="2">Hello everybody,<br>I'm UnCLe. Can Anybody fix to me regarding my problem.<br><br><u>Setup</u></font></div> <div><font face="Arial" size="2">IP Phone -- Asterisk with 4-port TDM FXO card -- Analog line x 4</font></div> <div> </div> <div><font face="Arial" size="2"><u>Problem 1</u></font></div> <div><font face="Arial" size="2">IP Phone call Mobile-A through TDM card analog line, Mobile-A ring and answer, both can hear and talk very fine, then IP Phone hang up. After few seconds, IP Phone wants to make call to Mobile-B, but when he/she pick up and listen to the headset, Mobile-A still online, which means the call don't hang up properly.</font></div> <div> </div> <div><font face="Arial" size="2"><u>Problem 2</u></font></div> <div><font face="Arial" size="2">1. Mobile call IP Phone</font></div> <div><font face="Arial" size="2">2. IP Phone ext 118 rings (as stated in
dialplan above)</font></div> <div><font face="Arial" size="2">3. After 15 seconds no answer from IP Phone, IP Phone stops ringing</font></div> <div><font face="Arial" size="2">4. After few seconds, IP Phone start rings again. <-- <font color="#ff0000">Here is the problem</font></font></div> <div> </div> <div><font face="Arial" size="2">To Mobile caller, he/she hear nothing weird except continuous ringin tone.</font></div> <div><br>here is the dialplan for your perusal, it is pretty staight forward btw;<br>>> <br>>> # extensions.conf<br>>> [from-pstn]<br>>> exten => s,1,Dial(SIP/118,15,)<br>>> exten => s,2,Playback(sorry) change=> s,3,Playback(vm-nobody...<br>>> exten => s,n,Playback(vm-nobodyavail) change=> s,4,Playback(vm-.....<br>>> exten => s,n,Playback(vm-goodbye) change=>
s,5,busy(3)<br>>> exten => s,n,Busy(3) change=> s,6,Hangup<br>>> exten => s,n,Hangup<br>>> <br>>> [from-internal]<br>>> exten => _9X.,1,DBput(RepeatDial/${CALLERIDNUM}=${EXTEN})<br>>> exten => _9X.,2,Dial(Zap/g1/${EXTEN:1})<br>>> exten => _9X.,3,Hangup<br>>> <br>>> # zaptel.conf<br>>> fxsks=1<br>>> fxsks=2<br>>> fxsks=3<br>>> fxsks=4<br>>> loadzone=us<br>>> defaultzone=us<br>>> <br>>> # zapata.conf (partial)<br>>> [channels]<br>>> language=en<br>>> context=from-pstn<br>>> signalling=fxs_ks<br>>> rxwink=300<br>>> callgroup=1<br>>> pickupgroup=1<br>>> group=1<br>>> callerid="89459722" <89459722><br>>> channel => 1<br>>> callerid="89459733"
<89459733><br>>> channel => 2<br>>> callerid="89412563" <89412563><br>>> channel => 3<br>>> ;callerid="78061817" <78061817><br>>> ;channel => 4<br> <br>Since we configured the * server use RealTime to handle SIP account, so <br>there is no sip.conf info attached.<br></div> <div><font face="Arial" size="2"><br>Hope the above example may explain the problem clearly and please do no hesitate to contact us if you need more info.</font></div> <div> </div><font face="Arial" size="2">That would be very much appreciated for your to get back to us with solution ASAP.</font> <div> </div> <div><font face="Arial" size="2">Regards,<br>UnCLe.<br></font></div></font></div><p> 
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