[asterisk-users] Problem Hangup

[FERNANDO VILLARROEL]

--- Knud Müller <k.mueller at portrix.net> wrote:

> FERNANDO VILLARROEL schrieb:
> > Hello list, i need help.
> >
> > My problem is that when I want to call (using ISDN
> > phone or internal SIP client) via the Sip provider
> a
> > real phone number (ISDN phone or internal SIP
> >
> >         Asterisk >> SIP ), I get a ring tone. When
> I
> > now decide to hang up (e.g. if 
> >
> > nobody answers), the called telephone continues to
> > ring almost forever.
> >
> > the sip.conf:
> >
> > [2563105]
> > accountcode = 2563105
> > username = 2563105
> > secret = 135
> > callerid = 412563105
> > context = test
> > canreinvite = no
> > dtmfmode = rfc2833
> > host = dynamic
> > insecure = very
> > language = es
> > nat = yes
> > qualify = yes
> > type = friend
> > disallow=all
> > allow=g729
> >
> > [nyphone]
> > accountcode=nyphone
> > canreinvite=no
> > reinvite=yes
> > username=test770
> > secret=test770
> > dtmfmode=rfc2833
> > host=72.55.143.XXX
> > insecure=very
> > language=es
> > nat=no
> > qualify=no
> > type=peer
> > disallow=all
> > allow=g729
> >
> > I attach sip debug one call.
> >
> > I use Asterisk 1.2.13
> >
> > I hope you understand me and help.
> >
> > Best regards
> >
> > Fernando Villarroel Noriel.
> > Chillan
> > Chile
> >
> > Sorry my English.
> >
> > 			
> > 	
> > 			
> > 	
> > 		
> > 	
> > 	
> > 		
> > 	
> >
> >
> >
> >        
> >
>
____________________________________________________________________________________
> > Looking for a deal? Find great prices on flights
> and hotels with Yahoo! FareChase.
> > http://farechase.yahoo.com/
> >
>
------------------------------------------------------------------------
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> If I got it right: you register to your SIP Provider
> which provides a 
> PSTN Number to you. You dial the PSTN Number which
> is forwarded to your 
> asterisk. Your asterisk dials the SIP phone
> (nyphone)?

Yes nyphone is my provider for everyone calls
internationational (prefix 00)

2563105 is one number provided for my Telco (E1) and
is one SIP client.

> Could you attach your dialplan?

exten => _00X.,1,dial(sip/${EXTEN:2}@nyphone,45)
exten => _00X.,2,hangup


the called telephone continues to ring almost forever.

> 
> Knud
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 



       
____________________________________________________________________________________
Get the free Yahoo! toolbar and rest assured with the added security of spyware protection.
http://new.toolbar.yahoo.com/toolbar/features/norton/index.php