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

[Josué Conti]

Hi Mike, all good? I thank its attention. Where I modify these parameters
that you said? Best Regards
Josué

2006/6/27, Mike Lynchfield <theclubvoip at gmail.com>:
>
> 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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