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

Horacio J. Peña horape at compendium.com.ar
Mon Nov 29 11:24:19 CST 2010 

On ISUP each channel is identified by a number called CIC, and the assignation
CIC <-> E1 timeslot is arbitrary. Your CIC 3 can as easily be channel 3 of your
first E1 as channel 27 of your 2nd E1 (I think it could even be a copper line or
a RTP stream, but let´s not go there)

If you establish a call on CIC 3 and you believe CIC 3 is the channel 3 of your
first E1 and the other side believes it´s the channel 4 of that same E1, you'll
get silence.

An easy test to see if there is a problem with the CICs is making several calls
at once. If you get audio, but from the wrong person it's surely a CIC mismatch,
and you can easily discover the right config.

Say that you do the calls:

A -> B, CIC 2
C -> D, CIC 3
E -> F, CIC 4

and when the calls are connected A is talking to B, that means that the channel
you believe to be CIC 2 is really CIC 3 (That would be the outcome if the other
side numbers the CIC from 1 as somebody suggested)

On Mon, Nov 29, 2010 at 06:48:44PM +0300, Timothy Smith wrote:
> 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,
> 
> -- 
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-- 
Horacio J. Peña
horape at compendium.com.ar
horape at uninet.edu

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