Dear Users, I seeking help on with the asterisk+libss7. the call is successfully setup but no audio either end.
I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2, chan_dahdi.c is too bing but i can send it if required(perhaps to add p->dialing = 0. I didnt do it correctly?) I appreciate your help in advance. Could someone please send me working confs/chan_dahdi.conf please! [r...@ivr asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] echocancel=yes echocancelwhenbridged=yes group=1 signalling=ss7 ss7type=itu ss7_called_nai=national ss7_calling_nai=national linkset=1 pointcode=25 adjpointcode=33 defaultdpc=33 networkindicator=national sigchan=1 cicbeginswith=2 channel=2-124 ss7_internationalprefix=000 ss7_nationalprefix=0 context=ss7 [r...@ivr1 asterisk]# cat /etc/dahdi/system.conf span=1,1,0,ccs,hdb3 bchan=2-31 mtp2=1 span=2,2,0,ccs,hdb3 bchan=32-62 span=3,3,0,ccs,hdb3 bchan=63-93 span=4,4,0,ccs,hdb3 bchan=94-124 loadzone = us defaultzone = us [r...@ivr asterisk]# Thank you! Kind Regards, -- _____________________________________________________________________ -- 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
