Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us
#zapata.conf
[trunkgroups]
[channels]
language=pt_BR
context=default
switchtype=qsig
pridialplan=private
prilocaldialplan=private
facilityenable = yes
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
immediate=no
callerid=asreceived
musiconhold=default
group=1
channel=>1-15
channel=>17-31
Best Regards
Josué
2006/6/27, Herchi Silviu <[EMAIL PROTECTED]>:
Hi,
Could you post your /etc/zaptel.conf and zapata.conf?
Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)?
Silviu
----
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me?Best Regards
Josué
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