[asterisk-ss7] no audio after update

[Rana Dhekial]

I have the following SVN versions and have audio.

 

[root at ildserver asterisk]# asterisk -rx "dahdi show version"
DAHDI Version: SVN-trunk-r6556 Echo Canceller:


[root at ildserver asterisk]# asterisk -rx "ss7 show version"
libss7 version: SVN-branch-1.0-r259M


[root at ildserver asterisk]# asterisk -rx "core show version"
Asterisk SVN-trunk-r191919M built by root @ ildserver on a i686 running Linux on 2009-04-29 18:26:27 UTC

 
> Date: Mon, 4 May 2009 14:19:58 -0300
> From: ngreco at gmail.com
> To: asterisk-ss7 at lists.digium.com
> Subject: Re: [asterisk-ss7] no audio after update
> 
> Do you have any news about this issue? I think I have the same problem
> using asterisk trunk + libss7-1.0.
> 
> Saludos,
> Nahuel Greco.
> 
> 
> 
> On Thu, Apr 30, 2009 at 1:38 PM, Matthew Fredrickson <creslin at digium.com> wrote:
> > Freddi Hansen wrote:
> >> Hi,
> >> I am using libss7.
> >> I have TE420 with ss7 link to a Huawei m800 on one side and SIP on the
> >> other side of a gateway
> >
> > I believe I know what is causing this problem.  Can you contact me
> > directly about this via IM, and I will see if we can resolve it.
> >
> > Matthew Fredrickson
> > Digium, Inc.
> >
> >>
> >> I have been using SVN-branch-1.6.0-r162806  and DAHDI Version: 2.0.0
> >> with no problems.
> >>
> >> I had a gateway out for service so I did upgrade to
> >> SVN-branch-1.6.0-r191214 and DAHDI Version: SVN-trunk-r6513.
> >>
> >> Now I can still call from ss7 side to SIP with no problems but I get no
> >> audio if I call from SIP to ss7.
> >>
> >> Has anyone else seen issues like this on latest versions.
> >>
> >> A snip from /etc/asterisk/chan_dahdi.conf
> >>
> >> ss7type = itu
> >> ss7_called_nai=International
> >> ss7_calling_nai=International
> >> linkset = 1
> >> pointcode = 0262
> >> adjpointcode = 3261
> >> defaultdpc = 3261
> >>
> >> cicbeginswith => 2
> >> networkindicator = national_spare
> >> sigchan => 1
> >> group => 1
> >> channel => 2-31
> >>
> >> and from /etc/dahdi/system.conf
> >> span=1,1,0,ccs,hdb3
> >> span=2,2,0,ccs,hdb3
> >> span=3,3,0,ccs,hdb3
> >> span=4,4,0,ccs,hdb3
> >>
> >>
> >> bchan=2-31
> >> dchan=1
> >>
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >>
> >> asterisk-ss7 mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-ss7

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