[asterisk-ss7] CIC Mapping chan_ss7

Thu Jun 1 02:27:26 MST 2006

I have my cic mapping corectly done and yet I have my calls dropping after
the first link.

The attached is my ss7 dump and ss7.conf

Thanks for you help

-----Original Message-----
From: asterisk-ss7-bounces at lists.digium.com
[mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Jacob Tinning
Sent: Wednesday, May 31, 2006 9:34 AM
To: asterisk-ss7 at lists.digium.com
Subject: Re: [asterisk-ss7] CIC Mapping chan_ss7

On Tue, 30 May 2006, leonimar cape wrote:

> I am getting silence/no audio on my second E1. I am
> using a A104D card. The called party was ringing but I
> only got silence when the call is answered. Can
> someone please help me.

I can try... I guess there is something wrong with your 'firstcic'
directives in ss7.conf.

> Below is my ss7.conf
>
>
> [linkset-siuc]
> enabled => yes
> enable_st => no
> use_connect => no
> hunting_policy => even_mru
> subservice => 8
> language => en
> context => outrt-003-SPRINT
> ;context => from-pstn
> t35 => 4000,st
> ;context => ext-did
>
> [link-l1]
> linkset => siuc
> channels => 1-15,17-31
> schannel => 16
> firstcic => 1
> ; echocancel: allways | 31speech | no
> echocancel=allways
> ; echocan_taps: 32 | 64 | 128 | 256
> echocan_taps=128
> ; echocan_train: between 10ms and 1000ms
> echocan_train=100
> enabled => yes
>
>
> [link-l2]
> linkset => siuc
> channels => 1-15,17-30
> schannel =>
> firstcic => 32

; It should be:
channels => 1-31
schannel =>
firstcic => 33

> ; echocancel: allways | 31speech | no
> ;echocancel=allways
> ; echocan_taps: 32 | 64 | 128 | 256
> ;echocan_taps=128
> ; echocan_train: between 10ms and 1000ms
> ;echocan_train=100
> enabled => yes
>
> [host-asterisk1.local]
> enabled => yes
> opc => 0x3
> dpc => siuc:0x28fe
> links => l1:1,l2:2

> --- leonimar cape <leo_mac_ph at yahoo.com> wrote:
>
>> Can please someone help on how does the CIC mapping of
>> the chan_ss7 works especially for configuring more
>> than 1 e1. Does it support skip on the timeslot 16?

>From my experience, it usually helps to write out all the cic's to
undestand the mapping.
The next 4 lines shows the cic's on 4 E1's. (The lines are long, so your
mail-reader will
probably ruin the alignment. To help you re-align, the pipe-chars '|' should
be vertically alligned.)

0  | 1  2  3   4   5   6   7   8   9  10  11  12  13  14  15  16  17  18  19
20  21  22  23  24  25  26  27  28  29  30  31
32 |33 34 35  36  37  38  39  40  41  42  43  44  45  46  47  48  49  50  51
52  53  54  55  56  57  58  59  60  61  62  63
64 |65 66 67  68  69  70  71  72  73  74  75  76  77  78  79  80  81  82  83
84  85  86  87  88  89  90  91  92  93  94  95
96 |97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115
116 117 118 119 120 121 122 123 124 125 126 127

The first column are the timeslot where a sync. signal is present.
The 'firstcic' directive should be the number in the second column (right
after the '|') 1, 33, 65 and 97.
Signalling timeslots are the column where the first '16' is in (16, 48, 80
and 112).

Usually you will only use 16 for signalling and then 48, 80 and 112 for
audio.

Now comes the tricky part.. The "channels" and "schannels" directive refers
to cic's at the
particular E1 so they should be less than or equal to 31 and more than or
equal to 1.

In contrary, the 'firstcic' directive refers to all the E1's so it should
typically be 1, 33, 65 or 97.

I hope this made it a little more clear for everybody on
<asterisk-ss7 at lists.digium.com>,
how to configure channel mappings for chan_ss7. I know channel and cic
mapping are not
the most easy subject in the world... (and Im not the best teacher in the
world).

>> I think my wrong CIC mapping is the reason why I get
>> silence on my second, third and forth e1. Also, may I
>> ask if the next release will support routing interms
>> of e1/t1 and not only by linkset? I think this will be
>> usefull specially if you are interconnecting to a
>> telco switch that uses trunk group for segregating
>> traffic.

No, the next version will not support routing. We (Sifira) does not need it
right now, so
we do not have plans to make it. However, anybody are very welcome to send
us a patch,
which includes routing :)

Mvh. Jacob

-- 
Jacob Tinning
System Developer                                           SIFIRA A/S
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-ss7
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ss7.conf
Type: application/octet-stream
Size: 1385 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060601/53108cdd/ss7-0001.obj
-------------- next part --------------
A non-text attachment was scrubbed...
Name: test.pcap
Type: application/octet-stream
Size: 18691 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060601/53108cdd/test-0001.obj