--- Knud Müller <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED],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 _______________________________________________ --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