hello from germany,
i'm using a TE110P in E1-mode with asterisk as a VOIP<>PSTN gateway. i
can
dial out with the sip-phones and everything is ok, but when i dial a wrong
phonenumber, with a normal phone i will hear a message telling that, but
asterisk passes no audio to the phone, like it worked with chan_capi and
isdn with "early B3 connect" enabled. the best would be to do all the
status-signalling like busytones etc. via audio like its done in grandmas
phone. does anyone know how to achieve this with zaptel?
thanks in advance!
dave
p.s.: my configfiles:
/etc/zaptel.conf:
--snip--
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = nl
defaultzone=nl
--snap--
zapata.conf:
--snip--
[channels]
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
internationalprefix=00
nationalprefix=0
usecallingpres=no
busydetect=no ; not need on pri
;callprogress=yes ; was yes but wiki says experimatley could be
produce ha
ngups
callwaitingcallerid=yes ; show callerid on callwaitingcalls
echotraining=no
echocancel=no
echocancelwhenbridged=no
overlapdial=no
;immediate=yes
;callerid=asreceived
callerid=no
language=de
rxgain=0.0
txgain=0.0
group=1
signalling=pri_cpe
context=default
channel => 1-15,17-31
--snap--
extensions.conf:
--snip--
[general]
static=yes
writeprotect=no
autofallthrough=yes
[default]
;;outbound via TE110P
exten => _0.,1,Dial(Zap/g1/${EXTEN},30)
exten => _0.,102,Busy
--snap--