[Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi again...

normally the 0/16 is a d-channel.

check the config in the zapata.conf. You should have
some thing like this

/etc/zapata.conf
bchan=1-15 
dchan=16 
bchan=17-31 

/etc/asterisk/zapata.conf
channel => 1-15,17-31 

i don't rember exactelly but in /proc/zaptel there is
the possibility to check if the channels are in use or
not. Maybe someone else can give you a hint...

sorry but i only interconnect Asterisk and H4K using
chan_capi and i have no experience with zapata ;-(

rich

--- Josué Conti <josueconti at gmail.com> wrote:

> Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386
> pri_dchannel: Ring requested
> on unconfigured channel 0/16 span 1
> I noticed this message in the CLI, when I tried to
> effect one call of HiPath
> 4000 for asterisk. Ring occurred, however when the
> voicemail of asterisk
> took care of call it was dumb, without no sound. I
> thank the attention
> Regards
> 
> Josué
> 
> 2006/6/26, Josué Conti <josueconti at gmail.com>:
> >
> >  Hi Richard.
> > Thank you very much for its attention. In the
> reality what is occurring is
> > that in some originated calls of the HiPath with
> destination to the Asterisk
> > they are being without the dumb and rings. I do
> not have this parameter in
> > my HiPath 4000, what I have seemed in the COT is
> TR6T (1tr6 isdn tie link)
> > would be this parameter?                          
>        Best Regards
> > Josué
> >
> > 2006/6/26, richard Coco <coco_richard at yahoo.com>:
> >
> > >
> > > Hi Josué
> > >
> > > if the Siemens phone calls Asterisk, it didn't
> get a
> > > dial tone from Asterisk? Is it correct?
> > >
> > > if yes, this is depending of Asterisk which
> didn't
> > > generates a ringback messages as it expexts dial
> ton
> > > generation localy. So try this workaround for
> HiPath
> > > local dial ton generation:
> > > -> Add option TR6Q(TRGT) to the class of trunk
> (COT)
> > > parameters
> > >
> > > hope it will help...
> > >
> > > rich
> > >
> > >
> > >
> > >
> > >
> > > --- Josué Conti <josueconti at gmail.com> wrote:
> > >
> > > >   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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