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

[richard Coco]

Hi again,

the TR6T parameter (i have german settings for my AMO
so it is TR6Q ;-)) resolved the same issue for my...
the difference is that i have an IP-trunk (using
oh323) between Asterisk and the HiPath. Have you tried
to remove the TR6T parameters...

Can you also paste the following outputs from the H4K

DISPLAY-APS:TYPE=PSGL,SP="y0*";
REGENERATE-TDCSU:PEN1=XX-XX-XX-XX; (change xx-xx-xx-xx
to the pin of the isdn trunk e.g 01-02-25-00) 
DISPLAY-COT:COTNO=XX; (change XX to the cot number of
the trunk)

if the log is not to huge please paste the last 30 min
of the history file. Try to reproduce the issue after
that type:

START-HISTA:RTYPE=SEARCHB,STIME="2006-06-27/09:00",ETIME="2006-06-27/09:30";

adjust the start time and the end time in a way that
the test is in the range between STIME and ETIME...

regards rich...

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

> 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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