do you have any relevant logs on asterisk console. set verbosity 3 unload chan_dahdi.so then load chan_dahdi.so you should see the ......cic expected on ........ logs. try to set that cic as cicbeginswith. On Mon, Nov 29, 2010 at 7:56 PM, Timothy Smith <timotsmith at gmail.com> wrote: > Thank you Gentlemen for your responses. > > I have done the dahdi_monitor, its only TX that has some input (see > sample output below). Thats for both outgoing and incoming calls. > > How can I verify the circuit mapping? My core engineer (telco company) > said that he is using the 1st channel for signalling and the rest for > voice. > > I appreciate your help. > > Tim > > [root at ivr asterisk]# dahdi_monitor 12 -vvv > > Visual Audio Levels. > -------------------- > Use chan_dahdi.conf file to adjust the gains if needed. > > ( # = Audio Level * = Max Audio Hit ) > <----------------(RX)----------------> > <----------------(TX)----------------> > ################### * > ^Ccntrl-c pressed 0) Tx: 2516 ( 3960) > ################# * > Rx: 0 ( 0) Tx: 3308 ( 3960)done cleaning up ... > exiting. > [root at ivr asterisk]# dahdi_monitor 3 -vvv > > Visual Audio Levels. > -------------------- > Use chan_dahdi.conf file to adjust the gains if needed. > > ( # = Audio Level * = Max Audio Hit ) > <----------------(RX)----------------> > <----------------(TX)----------------> > ########### * > ^Ccntrl-c pressed 0) Tx: 2111 ( 2790) > Rx: 0 ( 0) Tx: 2035 ( 2790)done cleaning up ... exiting. > [root at ivr asterisk]# > > > On Mon, Nov 29, 2010 at 4:57 PM, 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 > > > > -- > _____________________________________________________________________ > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101129/e39838ea/attachment.htm
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