---------------------------- Original Message
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Subject: [Fwd: Astersik+ss7]
From:    shaonopennets.net
Date:    Sat, May 19, 2007 10:00 am
To:      asterisk-ss7lists.digium.com
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Dear All,

I am using asterisk+ss7+TE407P card at my box. here i have 4
E1 and only
one signal channel. problem is that my 1st E1 work fine. but
from 2 to 4th
E1 i have got only ring tone but no voice from both side.
what is the
problem. pls suggest me. here is my conf file

zaptel.conf
---------------

span=1,1,2,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31

span=2,1,2,ccs,hdb3
bchan=33-63

span=3,1,2,ccs,hdb3
bchan=65-95
#bchan=1-31

span=4,1,2,ccs,hdb3
bchan=97-124


zapta.conf
-----------
adjpointcode = 1010
defaultdpc = 14671
networkindicator=national
; First signaling channel
sigchan = 16
; Begin CIC (Circuit indication codes) count with this
number

group =>1
cicbeginswith = 1
; Channels to associate with CICs on this linkset
channel => 1-15
; another cicbeginwith, so channel 16 is used for
signalling
cicbeginswith = 17
channel => 17-31

group => 2
cicbeginswith = 33
channel => 33-63

group => 3
cicbeginswith = 65
channel => 65-95

group => 4
cicbeginswith = 97
channel => 97-124

-----------------

pls suggest me

Best Regards

shaon


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Dear All

Can someone tell me what the logic behind the selection of
cic's for
outgoing calls is?

I have 2 E1's and for outgoing calls chan_ss7 should select
firstly, if
not in use, cic 62 and then 60, 58, 56, ... and so on.

But right now for example, I have 2 users calling
concurrently out and
on channels cic 56 and 58 and if I place a call then I hit
the cic 60.
why not 62, what is the logic behind this?

Thanks a lot

BR

Ercan





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