Hi Tim,
Make sure your CICs line up. Check that your T1-1 is the other side T1-1.
I had a similar problem and my CIC was not lined up. My T1-1 was their
T1-5.
Thanks,
Dave
-----Original Message-----
From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx
[mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Timothy Smith
Sent: Monday, November 29, 2010 9:57 AM
To: asterisk-ss7 at lists.digium.com
Subject: Re: Help with SS7 (No Audio)
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,
>>
>> --
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>
>
>
> --
> Regards,
> Abdul Basit | +92 32 1416 4196
>
> --
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