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

[Mike Lynchfield]

not sure. however pap2 devices usually have htat in regional stttings.. for
asterisk i don't know. im sales not tech ;)

sorry.. but you got a heads up

On 6/27/06, Josué Conti <josueconti at gmail.com> wrote:
>
> 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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-- 
Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.888.470.7253
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