Hello everybody, i?ve set up asterisk with an avm fritzcard pci, which is directly connected to my s0-bus from my mainline; so on this side i get incoming calls. On the otherside there are several sip-phones, which are to answer the incoming calls. This set up works quite well, when a call comes in. The problem starts when there comes a second (parallel) call on the s0-bus. Asterisk recognizes this call and dials for the sip-phones. They are ringing and if i answer the call with a second sip-phone, the audio from the first call isn?t transmitted anymore. The connection of the first call doesn?t break, but the audio is broken. I?m using Asterisk 1.0.9 in combination with chan_capi-cm 0.6 from sourceforge.net. Does anyone had the same problem? Is there any solutions to this? cheers, christian P.S.: Here is my configuration: capi.conf: [general] nationalprefixinternationalprefix=00 rxgain=0.8 txgain=0.8 [ISDN1] isdnmode=msn incomingmsn=<my_number> controller=1 group=1 softdtmf=on relaxdtmf=on accountcodecontext=incoming holdtype=hold echocancelold=yes devices=2 extensions.conf: [general] static=yes writeprotect=no [globals] PHONE12 => SIP/unertl PHONE13 => SIP/augustiner [incoming] exten => <my_number>,1,ChanIsAvail(${PHONE12}&${PHONE13}) exten => <my_number>,2,Dial(${PHONE12}&${PHONE13}) exten => <my_number>,3,HangUp exten => <my_number>,102,Answer exten => <my_number>,103,Playtones(busy) exten => <my_number>,104,Busy