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.
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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-usersIf 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)? Could you attach your dialplan? Knud
--- Knud M?ller <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