Hi,
I trying to use SS7 in loopback mode in my Asterisk box where a TE207P card
is installed. TE207P has two ports and I have a telco cross over cable
connected between port 1 and port 2.
"zap show status" indicates that both the ports are OK.
Description Alarms IRQ bpviol CRC4 Fra Codi
Options LBOT2XXP (PCI) Card 0 Span 1 OK 0 0 0
CCS HDB3 YEL 0 db (CSU)/0-133 feet (DSX-1)T2XXP (PCI) Card 0 Span 2
OK 0 0 0 CCS HDB3 YEL 0 db (CSU)/0-133 feet
(DSX-1)
zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
zapata.conf
[trunkgroups]
[channels]
group=1
signalling=ss7
ss7type = itu
context=from-outside
linkset = 1
pointcode = 1
adjpointcode =2
defaultdpc = 2
networkindicator=international
cicbeginswitch = 1
channel => 1-15
cicbeginswitch = 17
channel => 17-31
signchan = 16
; End of port 1 config
linkset=2
group =2
signalling =ss7
ss7type = itu
context =from-outside
pointcode = 2
adjpointcode = 1
defaultdpc = 1
networkindicator=international
cicbeginswitch = 1
channel = 32-46
cicbeginswitch = 17
channel = 48-62
signchan = 47
; End of port 2 config
When I call 201, from a SIP phone, it should go out using zap/g1, port 1 and
get looped back by the loopback cable and should come back to Asterisk through
port 2. But I get the following error message in the Asterisk console.
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/5551001-093ecf88",
"trunkdial,Zap/g1/201") in new stack
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/5551001-093ecf88", "Zap/g1/201") in
new stack
-- Called g1/201
[May 8 17:07:23] WARNING[5171]: chan_zap.c:9480 ss7_linkset: IAM on
unconfigured CIC 1
-- Hungup 'Zap/1-1'
-- No one is available to answer at this time (1:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Goto("SIP/5551001-093ecf88", "s-NOANSWER,1")
in new stack
-- Goto (macro-trunkdial,s-NOANSWER,1)
-- Executing [EMAIL PROTECTED]:1] Hangup("SIP/5551001-093ecf88", "") in new
stack
== Spawn extension (macro-trunkdial, s-NOANSWER, 1) exited non-zero on
'SIP/5551001-093ecf88' in macro 'trunkdial'
== Spawn extension (macro-trunkdial, s-NOANSWER, 1) exited non-zero on
'SIP/5551001-093ecf88'
[May 8 17:07:23] WARNING[5171]: chan_zap.c:9765 ss7_linkset: RLC on
unconfigured CIC 1
The Asterisk config ( sip.conf and extensions.conf ) should be fine as the same
call works when I configure the ports for PRI and connect loop back cable
between them. What am I doing wrong?
BTW, I successfully tested SS7 calls to and from an SS7 simulator using port 1.
thanks,
RD
_________________________________________________________________
With Windows Live for mobile, your contacts travel with you.
http://www.windowslive.com/mobile/overview.html?ocid=TXT_TAGLM_WL_Refresh_mobile_052008_______________________________________________
--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
