Hello. I have asterisk with an old avm b1 v3.0 configured and working with capi channel . The isdn card is connected to an S0 isdn bus of a siemens hipath 3000 version 4.0 It is possible to make outgoing calls and receive too, but I think that there is some kind of signaling problem when i call from an ip phone or soft ip phone because i do not get ring or busy tone calling to any PBX phone, but if i get outside line (i.e. calling a cell phone at PSTN) then i can hear the busy or ring tones. So, if i call to any pbx phone i do not hear anything until someone picks up the phone and if it is busy i do not know and after a while i get the normal call clearing and the call is finished. IP PHONE -----> B1 (ISDN) chan capi AT * ----------> S0 BUS HIPATH ---------> PSTN Does anybody know any useful trick to solve this? I know that may be it is a signaling problem and isdn related question, but if someone can help me it would be great. Pardon my bad English and thanks. Some configured parameters: **************************************************************************** ; ;capi.conf ; ; general section [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=es ; interface sections ... [ISDN1] ;ntmode=yes isdnmode=msn incomingmsn=620 controller=1 group=1 ;dialout group ;prefix=0 softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls holdtype=hold ; ;immediate=yes ;echosquelch=1 ;echocancel=yes ;echotail=64 bridge=yes callgroup=1 language=es *********************************************************************** ; ;indications.conf ; [general] country=es [es] description = Spain ringcadence =1500,3000 dial = 423 busy =425/200,0/200 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 ********************************************************************* ; ; part of dial extensions.conf for dialing outisde ;RDSI=620 is the number for the isdn S0 bus ; [capi-out] exten => _0.,1,NoOp("Salida a la calle" ${CALLERID}) exten => _0.,2,Dial(CAPI/g1/${RDSI},20) ... ********************************************************************** Thanks. Ricardo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060420/27c16552/attachment.htm