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Hi,<BR>
<BR>
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. <BR>
<BR>
"zap show status" indicates that both the ports are OK.<BR>
Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO<BR>T2XXP (PCI) Card 0 Span 1 OK 0 0 0 CCS HDB3 YEL 0 db (CSU)/0-133 feet (DSX-1)<BR>T2XXP (PCI) Card 0 Span 2 OK 0 0 0 CCS HDB3 YEL 0 db (CSU)/0-133 feet (DSX-1)<BR><BR>
zaptel.conf <BR>
<BR><FONT size=2>
span=1,1,0,ccs,hdb3<BR>
bchan=1-15,17-31<BR>
dchan=16<BR>
span=2,0,0,ccs,hdb3<BR>
bchan=32-46,48-62<BR>
dchan=47<BR>
<BR>
zapata.conf<BR>
<BR>
[trunkgroups]<BR>
[channels]<BR>
group=1<BR>
signalling=ss7<BR>
ss7type = itu<BR>
context=from-outside<BR>
linkset = 1<BR>
pointcode = 1<BR>
adjpointcode =2<BR>
defaultdpc = 2<BR>
networkindicator=international<BR>
cicbeginswitch = 1<BR>
channel => 1-15<BR>
cicbeginswitch = 17<BR>
channel => 17-31<BR>
signchan = 16<BR>
<BR>
; End of port 1 config<BR>
<BR>
linkset=2<BR>
group =2<BR>
signalling =ss7<BR>
ss7type = itu<BR>
context =from-outside<BR>
pointcode = 2<BR>
adjpointcode = 1<BR>
defaultdpc = 1<BR>
networkindicator=international<BR>
cicbeginswitch = 1<BR>
channel = 32-46<BR>
cicbeginswitch = 17<BR>
channel = 48-62<BR>
signchan = 47<BR>
<BR>
; End of port 2 config<BR>
<BR>
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.<BR>
<BR>
== Using SIP RTP CoS mark 5<BR>
-- Executing [201@from-inside:1] Macro("SIP/5551001-093ecf88", "trunkdial,Zap/g1/201") in new stack<BR>
-- Executing [s@macro-trunkdial:1] Dial("SIP/5551001-093ecf88", "Zap/g1/201") in new stack<BR>
-- Called g1/201<BR>
[May 8 17:07:23] WARNING[5171]: chan_zap.c:9480 ss7_linkset: IAM on unconfigured CIC 1<BR>
-- Hungup 'Zap/1-1'<BR>
-- No one is available to answer at this time (1:0/0/0)<BR>
-- Executing [s@macro-trunkdial:2] Goto("SIP/5551001-093ecf88", "s-NOANSWER,1") in new stack<BR>
-- Goto (macro-trunkdial,s-NOANSWER,1)<BR>
-- Executing [s-NOANSWER@macro-trunkdial:1] Hangup("SIP/5551001-093ecf88", "") in new stack<BR>
== Spawn extension (macro-trunkdial, s-NOANSWER, 1) exited non-zero on 'SIP/5551001-093ecf88' in macro 'trunkdial'<BR>
== Spawn extension (macro-trunkdial, s-NOANSWER, 1) exited non-zero on 'SIP/5551001-093ecf88'<BR>
[May 8 17:07:23] WARNING[5171]: chan_zap.c:9765 ss7_linkset: RLC on unconfigured CIC 1<BR>
<BR>
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?<BR>
<BR>
BTW, I successfully tested SS7 calls to and from an SS7 simulator using port 1. <BR>
<BR>
<BR>
thanks,<BR>
<BR>
<BR>
RD<BR></FONT><br /><hr />With Windows Live for mobile, your contacts travel with you. <a href='http://www.windowslive.com/mobile/overview.html?ocid=TXT_TAGLM_WL_Refresh_mobile_052008' target='_new'>Connect on the go.</a></body>
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