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Never mind. I should use "cicbeginswith" not "cicbeginswit<U><STRONG>c</STRONG></U>h" and "sigchan" not "sig<U><STRONG>n</STRONG></U>chan" in the zapata.conf. Loopback between port 1 and 2 works now.<BR><BR>
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From: dhekial@msn.com<BR>To: asterisk-ss7@lists.digium.com<BR>Date: Fri, 9 May 2008 12:39:09 -0700<BR>Subject: [asterisk-ss7] SS7 with port 1 and port 2 in loopback<BR><BR>
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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>
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