Unwanted bridging of sip channels

managed to succesfully register all the lines from my fxo-4 wellgate into asterisk, then I can place and receive calls(setting the hot line function to an asterisk extension)… but…
The problem arises when I try to place two simultaneous, inbound or outbound calls, for some reason asterisk is mixing the channels and half hanging one. Sometimes my asterisk box is crashing after.
Settings in the wellgate:
Mode = Proxy
Detect silence = no (setting only available through telnet connection)
Echo canceller= yes
codec = ulaw
dtmf = inband
routes = From ip to FXO ---- From FXO to IP

In asterisk:
add peers for every line (put an x in line number in the wellgate if you don’t want the to register this line)

;this is for receiving calls
exten => 901,hint,SIP/joe&SIP/silv&SIP/joe_office ; Channel hints for presence
exten => 901,1,answer
exten => 901,2,wait,1
exten => 901,3,PlayBack(Thankyouktob)
exten => 901,4,NoOp(${CALLERID})
exten => 901,5,Dial(${HINT},30)
exten => 901,6,Playback(vm-nobodyavail)
exten => 901,7,Playback(vm-intro)
exten => 901,8,voicemail(s500)
exten => 901,106,Playback(vm-nobodyavail)
exten => 901,107,Playback(vm-intro)
exten => 901,108,voicemail(s500)
exten => 901,109,Hangup()
same as above but replacing the extension for 902

With this configuration if an extension is busy and a second call enters, to the available fxo port, all my phones ring ok, but if I answer any of them, the channels are bridged and the mess starts.
With outbound calls the behavior is similar, if I select line one by the command dial(SIP/linenumber1) I hear tone from the fxo and can place a call normally, but if try to connect to the second outbound line dial(SIP/linenumber2), it jumps to busy and half hangs the first line.

Hope this helps some one, and I will appreciate any tips about why my sip channels are bridging.

Try asking on the asterisk users list:
You can learn a lot from there.