We have gone round and round on getting our ss7 link up. We can get the cics to align and the signaling link to come up. However, when we dial there is no audio in either direction.
Chan_dahdi: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes threewaycalling=no transfer=yes canpark=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 immediate=no prematureaudio=no language=en group=1 signalling = ss7 ss7type = itu linkset = 1 pointcode=6314 ; switch point code adjpointcode=12450 ; peer point code. defaultdpc=12450 ; per point code. networkindicator=international slc=0 ;ss7_internationalprefix = 00 ;ss7_nationalprefix = 0 ;ss7_subscriberprefix = ;ss7_unknownprefix = mtp2=1 sigchan=1 context=default cicbeginswith = 2 channel = 2-31 cicbeginswith = 33 channel = 32-62 cicbeginswith = 65 channel = 63-93 cicbeginswith = 97 channel = 94-124 Dahdi system.conf span=1,1,0,ccs,hdb3 bchan=2-31 dchan=1 echocanceller=mg2,2-31 span=2,0,0,ccs,hdb3 bchan=32-62 echocanceller=mg2,32-62 span=3,0,0,ccs,hdb3 bchan=63-93 echocanceller=mg2,63-93 span=4,0,0,ccs,hdb3 bchan=94-124 echocanceller=mg2,94-124 loadzone = fr defaultzone = fr Any ideas? Running Asterisk 1.8.4.4, DAHDI Version: 2.4.1.2 Echo Canceller: MG2, libss7 version: 1.0.2 -- -- _____________________________________________________________________ -- 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
