Gustavo,

Thanks for your response.

I make the configuration like you recommend, and the problem persist.
With the first E1, the call is ok, but with the least E1, the call is completed, but doesn't have audio. Other information is, the asterisk cannot create linkset 2 and 3 because this linksets doesn't have sigchan and when I place a call using these channels, asterisk crash with a segment fault.
I'm with asterisk 1.8.6.0 and libss7 1.0.2.

Regards,

Rodrigo Passos


Em 31/10/2011 22:59, Gustavo Mársico escreveu:
Just to mention, Alcatel S12 and Neax 61E aren't softswitches, they're good old 
fashion TDM switches.

If you need to ITX with 4 different switches you will need at least 4 SPC, and 
you defined just 2 (80 as STP and 100 as SP).

Guessing that S12_01 and NEAX_02 are acting as STP for S12_02 and NEAX_01, and 
assuming S12_01 is 80, S12_02 is 100, NEAX_01 is 120 and NEAX_02 is 140, it 
should be something like this:

;s12_01
;because you've cic's with S12_01
linkset=1
pointcode=80
adjpointcode=80

cicbeginswith = 1
networkindicator=national
sigchan = 16
channel = 1-15,17-31

;s12_02
;just cics
linkset=2
pointcode=100
adjpointcode=80                         'STP s12_01
defaultdpc=80

cicbeginswith = 1
networkindicator=national
channel = 32-62

;neax_01
;just cics
linkset=3
pointcode=120
adjpointcode=140                        'STP neax_02
defaultdpc=140

cicbeginswith = 1
networkindicator=national
channel = 63-93

;neax_02
;because you've cic's with neax_02
linkset=4
pointcode=140
adjpointcode=140

cicbeginswith = 1
networkindicator=national
sigchan = 109
channel = 94-108,110-124


In any case, you can use 1-31 as CIC in that scenario, you just need to get the 
proper information to define the linksets.

Regards

Gustavo


On Oct 31, 2011, at 8:13 PM, Rodrigo Ricardo Passos wrote:

Hi all,

I have the following scenario:

The telco company uses 4 different Softswitchs to compose my SS7 
interconnection, so 4 E1s to have redundancy.  Each this one uses channels 1 up 
to 31. The map of the firsts is equal the ID of the asterisk channels, but the 
next ID´s isn't the same and the signaling doesn't work. The telco cannot 
change the CIC configuration to have the same CIC in my configuration. I have 
one TE405P. The only way to solve this problem is change the CIC in the telco 
company or i can change CIC maps in my asterisk box? Only the first E1 align 
with the first softswitch.  When I a place a call using channel 64 of my third 
E1, telco doesn't have CIC 64, but have CIC 2 and the cannot be complete 
because the CIC isn't the same. What is the solution?

First E1:  Asterisk: 1 - 31 (16 signaling) -  Alcatel: S12_01 1-31 (16 
signaling)
Second E1 Asterisk: 32 - 62 (no signaling - voice only) - Alcatel: S12_02 1 - 31
Third E1: Asterisk: 63 - 93 (no signaling - voice only) - NEC: NEAX_01  1 - 31
Fourth E1: Asterisk: 94 - 124 (16 signaling) - NEC: NEAX_02 1 - 31

All dpcs are different; each this one have an unique ID for each E1, only 
problem is the signaling.
My configurations are:

system.conf:
span=1,1,0,ccs,hdb3
# termtype: unknown
bchan=1-15,17-31
mtp2=16

# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS
span=2,1,0,ccs,hdb3
# termtype: unknown
bchan=32-62

# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS
span=3,1,0,ccs,hdb3
# termtype: unknown
bchan=63-93

# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS
span=4,1,0,ccs,hdb3
# termtype: unknown
bchan=94-107,110-124
mtp2=108

# Global data

loadzone        = us
defaultzone     = us


chan_dahdi.conf:

[trunkgroups]

[channels]
context=interconexoes
ss7type=itu
signalling=ss7
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
networkindicator=national
echotraining=yes
echotraining=800
echocancel=yes

group=1

linkset=1
pointcode=100
defaultdpc=80
adjpointcode=80

cicbeginswith=1
channel=1-15
cicbeginswith=17
channel=17-31
sigchan=16

cicbeginswith=32
channel=32-62

pointcode=100
defaultdpc=90
adjpointcode=90

cicbeginswith=63
channel=63-93

cicbeginswith=94
channel=94-107
cicbeginswith=110
channel=110-124
sigchan=108





--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-ss7

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
    http://lists.digium.com/mailman/listinfo/asterisk-ss7


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-ss7

Reply via email to