[Asterisk-Dev] Sipura 3000 Disconnect Singnel

Kaloyan Kovachev kkovachev at varna.net
Mon Dec 5 02:37:44 MST 2005


Hi,
 it is safe to (and you should) hangup if your DeadAGI is executed from the h
extension. In my dialplan i am using something like this:

 exten => 111,1,Answer
 exten => 111,2,AGI(authenticate_and_get_the_number_to_dial)
 exten => 111,3,Dial(${TRUNK}/${DIAL_NUM}|${WAIT_TIME}|ghH)   ; can be done
from AGI directly
 exten => 111,4,NoOp(closed from the remote party)

 exten => h,1,DeadAGI(write_cdr|pass_CDR_fields_we_need)  ; if you pass the
fields to AGI it is safer than reading them from AGI itself
 exten => h,2,Congestion(if_closed_from_remote_only)   ; this is not executed
if the first party hangup too, but 'tells' that the call is ended otherwise

This won't fix the problem with hangup from Sipura (phone) side, but you can
do this with '*' now (hH Dial parameters)

Hope this helps

On Mon, 05 Dec 2005 12:20:07 +0545, Abdul Lateef Khan wrote
> Hi,
> 
> I am trying in Doha Qatar, and i am not able to find such as option 
> inside to hangup the phone when second party closed the phone.
> 
> I am using call forwarding from sipura to one of my extention using 
> 111 phone number. after that the perl agi script will authunticate 
> user and forward him to the terminator. while the call is connected 
> i can see two active channel one for Sipura to Asterisk using 111 
> number and second is Asterisk to terminator using destination number.
> 
> But when destination hangup the phone, the active channel of 
> destination became disapear but the call from sipura to Asterisk 
> using 111 number continue, till i remove the PSTN cable from Sipura.
> 
> And i am not using $AGI->hangup(); becuase i am using DeadAGI to 
> calculate the CDR etc..
> 
> I hope this little explination do more clear.
> 
> >From: Rich Adamson  <radamson at routers.com>
> >Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> >To: Asterisk Developers Mailing List  <asterisk-dev at lists.digium.com>
> >Subject: Re: [Asterisk-Dev] Sipura 3000 Disconnect Singnel
> >Date: Sun,  4 Dec 2005 07:21:34 -0600
> 
> >Not sure what you mean by "fxo phone", but if that really means a pstn 
> >user's
> >phone, then it sounds like the problem is with the spa3k detecting 
> >disconnect
> >supervision.
> >
> >In the US, disconnect supervision is generally a momentary tip-ring open
> >(eg, no voltage), and the spa3k will detect that. Since you didn't mention
> >which country you're trying to do this in, you will need to determine what
> >the disconnect supervision is in use (if any) and see if there is a 
> >parameter
> >within the spa3k to handle it.
> >
> >If you can't find anything relative to disconnect supervision for your 
> >area,
> >then put a voltmeter across tip-ring and see what happens when the phone
> >hangs up. Some countries use tones for disconnect supervision which
> >obviously you won't see with a voltmeter.
> >
> >
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