[asterisk-ss7] SS7 with port 1 and port 2 in loopback

[Rana Dhekial]

         Never mind.  I should use "cicbeginswith" not "cicbeginswitch" and "sigchan" not "signchan" in the zapata.conf. Loopback between port 1 and 2 works now.


From: dhekial at msn.comTo: asterisk-ss7 at lists.digium.comDate: Fri, 9 May 2008 12:39:09 -0700Subject: [asterisk-ss7] SS7 with port 1 and port 2 in loopback


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,hdb3bchan=1-15,17-31dchan=16span=2,0,0,ccs,hdb3bchan=32-46,48-62dchan=47 zapata.conf [trunkgroups][channels]group=1signalling=ss7ss7type = itucontext=from-outsidelinkset = 1pointcode = 1adjpointcode =2defaultdpc = 2networkindicator=internationalcicbeginswitch = 1channel => 1-15cicbeginswitch = 17channel => 17-31signchan = 16 ;   End of port 1 config linkset=2group =2signalling =ss7ss7type = itucontext =from-outsidepointcode = 2adjpointcode = 1defaultdpc = 1networkindicator=internationalcicbeginswitch = 1channel = 32-46cicbeginswitch = 17channel = 48-62signchan = 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 [201 at from-inside:1] Macro("SIP/5551001-093ecf88", "trunkdial,Zap/g1/201") in new stack-- Executing [s at macro-trunkdial: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 [s at macro-trunkdial:2] Goto("SIP/5551001-093ecf88", "s-NOANSWER,1") in new stack-- Goto (macro-trunkdial,s-NOANSWER,1)-- Executing [s-NOANSWER at macro-trunkdial: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

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