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 >
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