[Asterisk-Users] Re: Asterisk x Siemens HiPath 4000

[Josué Conti]

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

#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 <Silviu.Herchi at arcelor.com>:
>
>  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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