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

[Mike Lynchfield]

HERE IS answer !@!


had same problem..

make the settings for 90 volt.. not 70 volt ringer..

make it trapezoidal not sinusoisal

make it 900 ohm not 600 impedence..

that worked for pap2's

seem siemens are made for europe style ring voltage not north american.




On 6/27/06, Herchi Silviu <Silviu.Herchi at arcelor.com> wrote:
>
>  Hello,
>
> The main differences I can see:
>
> - in zaptel.conf
> you have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a
> timer for the PBX - on my setup the PBX is the master clock and Asterisk is
> the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4
> error correction, my setup is span=1,1,0,ccs,hdb3,crc4)
>
> - in zapata.conf
> I have switchtype=EuroISDN. Generally speaking, try using other
> switchtypes.
>
> Regards,
>
> Silviu
>  ------------------------------
>  *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Josué Conti
> *Sent:* 27 June 2006 14:41
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000
>
>  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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> >
>
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-- 
Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.888.470.7253
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