Any particularly reason why your CIC begins with 2? I think they usually starts at 1. Check with the other side of the connection on CIC configurations. On Mon, Nov 29, 2010 at 8:57 AM, Abdul Basit <basit.engg at gmail.com> wrote: > Try sending a call via call file and see if you are getting both call legs. > > callchannel.sh > > #!/bin/bash > echo "Channel: DAHDI/$1/$2 > Callerid: $2 > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: ss7 > Application: Echo" > /var/spool/asterisk/tmp/test.call > > mv /var/spool/asterisk/tmp/test.call /var/spool/asterisk/outgoing > > dahdi_monitor $1 -vv > > This is the way i verify the call legs. > > chmod +x callchannel.sh > > ./callchannel.sh channelNumber someNumber > ./callchannel.sh 3 123456789 > > > Most of the time problem is cic miss-match. > I hope this will help you debugging the issue. > > > > On Mon, Nov 29, 2010 at 6:34 PM, Timothy Smith <timotsmith at gmail.com>wrote: > >> 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! >> >> [root at 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 >> [root at 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 >> [root at 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 >> > > > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 > > -- > _____________________________________________________________________ > -- 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 > -- Best regards, Tuan Le -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101129/e132bc52/attachment.htm
|