[asterisk-ss7] Help with SS7 (No Audio)

[Abdul Basit]

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