FERNANDO VILLARROEL schrieb: > --- 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. > So its an outbound call not an inbound call! > >> Could you attach your dialplan? >> > > exten => _00X.,1,dial(sip/${EXTEN:[EMAIL PROTECTED],45) > exten => _00X.,2,hangup > This looks OK. I'd recommend to record the SIP communication with your provider. Do on the CLI: "SIP debug". You should see after latest after 45 seconds that asterisk sends an hangup request to your provider. If * sends the request it must be your provider. If it is not sent, something is wrong with your *. > > 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 > >
-- Knud A. Müller Geschäftsführer Tel.: 040/398053-11 Fax: 040/398053-29 e-Mail: [EMAIL PROTECTED] portrix.net GmbH Stresemannstr. 375 22761 Hamburg HRB 79850 (Amtsgericht Hamburg) Geschäftsführer: Knud Alex Müller, Henning Voss, Niclas Schroeder http://www.portrix.net _______________________________________________ --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