[asterisk-dev] Re: regarding TDM FXO Hangup problem

sword power swordpow at yahoo.com
Mon Aug 28 21:27:39 MST 2006


 Hello everybody,
I'm UnCLe. Can Anybody fix to  me regarding my problem.

Setup
 IP Phone -- Asterisk with 4-port TDM FXO card --  Analog line x 4
  
 Problem 1
 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.
  
 Problem 2
 1. Mobile call IP Phone
 2. IP Phone ext 118 rings (as stated in dialplan  above)
 3. After 15 seconds no answer from IP Phone, IP  Phone stops ringing
 4. After few seconds, IP Phone start rings again.  <-- Here is the problem
  
 To Mobile caller, he/she hear nothing weird except  continuous ringin tone.
 
here is the dialplan for your perusal, it is pretty staight forward  btw;
>> 
>> # extensions.conf
>>  [from-pstn]
>> exten => s,1,Dial(SIP/118,15,)
>> exten  => s,2,Playback(sorry)            change=>  s,3,Playback(vm-nobody...
>> exten => s,n,Playback(vm-nobodyavail)    change=> s,4,Playback(vm-.....
>> exten =>  s,n,Playback(vm-goodbye)       change=> s,5,busy(3)
>> exten =>  s,n,Busy(3)                    change=> s,6,Hangup
>> exten =>  s,n,Hangup
>> 
>> [from-internal]
>> exten =>  _9X.,1,DBput(RepeatDial/${CALLERIDNUM}=${EXTEN})
>> exten =>  _9X.,2,Dial(Zap/g1/${EXTEN:1})
>> exten => _9X.,3,Hangup
>>  
>> # zaptel.conf
>> fxsks=1
>> fxsks=2
>>  fxsks=3
>> fxsks=4
>> loadzone=us
>>  defaultzone=us
>> 
>> # zapata.conf (partial)
>>  [channels]
>> language=en
>> context=from-pstn
>>  signalling=fxs_ks
>> rxwink=300
>> callgroup=1
>>  pickupgroup=1
>> group=1
>> callerid="89459722"  <89459722>
>> channel => 1
>> callerid="89459733"  <89459733>
>> channel => 2
>> callerid="89412563"  <89412563>
>> channel => 3
>> ;callerid="78061817"  <78061817>
>> ;channel => 4
 
Since we configured the *  server use RealTime to handle SIP account, so 
there is no sip.conf info  attached.

 
Hope the above example may explain the problem  clearly and please do no hesitate to contact us if you need more  info.
  
That would be very much appreciated for your  to get back to us with solution ASAP.   
 Regards,
UnCLe.



 		
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