do you have any relevant logs on asterisk console.<div><br></div><div>set verbosity 3 </div><div>unload chan_dahdi.so</div><div><br></div><div>then load chan_dahdi.so</div><div> </div><div>you should see the ......cic expected on ........ logs.</div>
<div><br></div><div>try to set that cic as <span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; color: rgb(80, 0, 80); ">cicbeginswith</span>.</div><div><br></div>
<div><br><br><div class="gmail_quote">On Mon, Nov 29, 2010 at 7:56 PM, Timothy Smith <span dir="ltr"><<a href="mailto:timotsmith@gmail.com">timotsmith@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Thank you Gentlemen for your responses.<br>
<br>
I have done the dahdi_monitor, its only TX that has some input (see<br>
sample output below). Thats for both outgoing and incoming calls.<br>
<br>
How can I verify the circuit mapping? My core engineer (telco company)<br>
said that he is using the 1st channel for signalling and the rest for<br>
voice.<br>
<br>
I appreciate your help.<br>
<br>
Tim<br>
<br>
[root@ivr asterisk]# dahdi_monitor 12 -vvv<br>
<br>
Visual Audio Levels.<br>
--------------------<br>
Use chan_dahdi.conf file to adjust the gains if needed.<br>
<br>
( # = Audio Level * = Max Audio Hit )<br>
<----------------(RX)----------------> <----------------(TX)----------------><br>
################### *<br>
^Ccntrl-c pressed 0) Tx: 2516 ( 3960)<br>
################# *<br>
Rx: 0 ( 0) Tx: 3308 ( 3960)done cleaning up ...<br>
exiting.<br>
[root@ivr asterisk]# dahdi_monitor 3 -vvv<br>
<br>
Visual Audio Levels.<br>
--------------------<br>
Use chan_dahdi.conf file to adjust the gains if needed.<br>
<br>
( # = Audio Level * = Max Audio Hit )<br>
<----------------(RX)----------------> <----------------(TX)----------------><br>
########### *<br>
^Ccntrl-c pressed 0) Tx: 2111 ( 2790)<br>
Rx: 0 ( 0) Tx: 2035 ( 2790)done cleaning up ... exiting.<br>
[root@ivr asterisk]#<br>
<div class="im"><br>
<br>
On Mon, Nov 29, 2010 at 4:57 PM, Abdul Basit <<a href="mailto:basit.engg@gmail.com">basit.engg@gmail.com</a>> wrote:<br>
</div><div><div></div><div class="h5">> Try sending a call via call file and see if you are getting both call legs.<br>
> callchannel.sh<br>
> #!/bin/bash<br>
> echo "Channel: DAHDI/$1/$2<br>
> Callerid: $2<br>
> MaxRetries: 2<br>
> RetryTime: 60<br>
> WaitTime: 30<br>
> Context: ss7<br>
> Application: Echo" > /var/spool/asterisk/tmp/test.call<br>
> mv /var/spool/asterisk/tmp/test.call /var/spool/asterisk/outgoing<br>
> dahdi_monitor $1 -vv<br>
> This is the way i verify the call legs.<br>
> chmod +x callchannel.sh<br>
> ./callchannel.sh channelNumber someNumber<br>
> ./callchannel.sh 3 123456789<br>
><br>
> Most of the time problem is cic miss-match.<br>
> I hope this will help you debugging the issue.<br>
><br>
><br>
> On Mon, Nov 29, 2010 at 6:34 PM, Timothy Smith <<a href="mailto:timotsmith@gmail.com">timotsmith@gmail.com</a>> wrote:<br>
>><br>
>> Dear Users,<br>
>><br>
>> I seeking help on with the asterisk+libss7. the call is successfully<br>
>> setup but no audio either end.<br>
>><br>
>> I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2,<br>
>> chan_dahdi.c is too bing but i can send it if required(perhaps to add<br>
>> p->dialing = 0. I didnt do it<br>
>> correctly?)<br>
>><br>
>> I appreciate your help in advance. Could someone please send me<br>
>> working confs/chan_dahdi.conf please!<br>
>><br>
>> [root@ivr asterisk]# cat chan_dahdi.conf<br>
>> [trunkgroups]<br>
>> [channels]<br>
>> echocancel=yes<br>
>> echocancelwhenbridged=yes<br>
>> group=1<br>
>> signalling=ss7<br>
>> ss7type=itu<br>
>> ss7_called_nai=national<br>
>> ss7_calling_nai=national<br>
>> linkset=1<br>
>> pointcode=25<br>
>> adjpointcode=33<br>
>> defaultdpc=33<br>
>> networkindicator=national<br>
>> sigchan=1<br>
>> cicbeginswith=2<br>
>> channel=2-124<br>
>> ss7_internationalprefix=000<br>
>> ss7_nationalprefix=0<br>
>> context=ss7<br>
>> [root@ivr1 asterisk]# cat /etc/dahdi/system.conf<br>
>> span=1,1,0,ccs,hdb3<br>
>> bchan=2-31<br>
>> mtp2=1<br>
>> span=2,2,0,ccs,hdb3<br>
>> bchan=32-62<br>
>> span=3,3,0,ccs,hdb3<br>
>> bchan=63-93<br>
>> span=4,4,0,ccs,hdb3<br>
>> bchan=94-124<br>
>><br>
>> loadzone = us<br>
>> defaultzone = us<br>
>> [root@ivr asterisk]#<br>
>><br>
>><br>
>> Thank you!<br>
>> Kind Regards,<br>
>><br>
>> --<br>
>> _____________________________________________________________________<br>
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>><br>
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>> To UNSUBSCRIBE or update options visit:<br>
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><br>
><br>
><br>
> --<br>
> Regards,<br>
> Abdul Basit | +92 32 1416 4196<br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
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><br>
<br>
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_____________________________________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br><div>Regards,</div><br>Abdul Basit | +92 32 1416 4196<br>
</div>