Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy analog extensions.
my dial plan is like callers dial into asterisk(span1) and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....the prob is when people call and when asterisk attempts to transfer the call, the call dconnects with CHANUNVAIL error... >>> ZAPTEL.CONF span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 >>> ZAPATA.CONF context=pri-pstn switchtype=euroisdn pridialplan=local usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 immediate=yes musiconhold=default signalling = pri_cpe channel => 1-15 channel => 17-31 context=pri-legacy immediate=yes group=2 overlapdial=yes signalling = pri_net channel => 32-46 channel => 48-62>>> EXTENSIONS.CONF ; ; Context PRI-Public ; [pri-pstn] ; include => default ; exten => s,1,Answer exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx exten => s,3,Hangup ; ; Context PRI-legacy ; [pri-legacy] ; include => default ; exten => s,1,Answer exten => s,2,DigitTimeout,2 exten => s,3,ResponseTimeout,2 exten => _X.,1,Dial(Zap/g1/${EXTEN}) exten => _X.,2,Congestion
_______________________________________________ -- 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