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: [EMAIL PROTECTED]: [EMAIL PROTECTED]: 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 [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

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