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

[Timothy Smith]

Thank Dave do your advise.

Please advise me further, how do I verify the CICs and T1-1 (do u mean
time slots?) are line up correctly? I can meet the telco engineer but
need to explain to him properly and make my point. Unfornately, he
doesnt know asterisk :( (only knows his huawei switch).

By the way, i forgot to mention, when I turn on crc4 (in my
dahdi/system.conf), the link just starts coming up and down every
time! (see output below)

[Nov 29 10:47:45] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2
link down (SLC 0)
MTP2 link up (SLC 0)
[Nov 29 10:47:48] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2
link down (SLC 0)
MTP2 link up (SLC 0)
[Nov 29 10:47:50] WARNING[7026]: chan_dahdi.c:9974 ss7_linkset: MTP2
link down (SLC 0)
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting
retransmission


Thanks,
Tim


On Mon, Nov 29, 2010 at 6:40 PM, dave george <dgeorge at teletoneinc.com> wrote:
> 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 at lists.digium.com
> [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: [asterisk-ss7] 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,