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
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