[asterisk-users] [ [asterisk-ss7] libss7 2asterisk box

[aymen warfalli]

Hi list 
 
I plan to connect two asterisk box using libss7 ,i read the list messages ( thanks for this great jop) , i installed all the packegs with digium  single E1 link in both boxes with cenos 5 and every thing is looking ok excact when i am trying to call using sip channel it shows some problems here is muy configrations file  
 server A--B
 
zaptel.conf
span=1,0,0,ccs,hdb3  ;span=1,1,0,ccs,hdb3  server B 
bchan=1-15,17-31  
dchan=16
loadzone = us
defaultzone = us
ztcfg -vv
Zaptel Version: SVN--rEcho Canceller: MG2Configuration======================
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)Channel 02: Clear channel (Default) (Slaves: 02)Channel 03: Clear channel (Default) (Slaves: 03)Channel 04: Clear channel (Default) (Slaves: 04)Channel 05: Clear channel (Default) (Slaves: 05)Channel 06: Clear channel (Default) (Slaves: 06)Channel 07: Clear channel (Default) (Slaves: 07)Channel 08: Clear channel (Default) (Slaves: 08)Channel 09: Clear channel (Default) (Slaves: 09)Channel 10: Clear channel (Default) (Slaves: 10)Channel 11: Clear channel (Default) (Slaves: 11)Channel 12: Clear channel (Default) (Slaves: 12)Channel 13: Clear channel (Default) (Slaves: 13)Channel 14: Clear channel (Default) (Slaves: 14)Channel 15: Clear channel (Default) (Slaves: 15)Channel 16: D-channel (Default) (Slaves: 16)Channel 17: Clear channel (Default) (Slaves: 17)Channel 18: Clear channel (Default) (Slaves: 18)Channel 19: Clear channel (Default) (Slaves: 19)Channel 20: Clear channel (Default) (Slaves: 20)Channel 21: Clear channel (Default) (Slaves: 21)Channel 22: Clear channel (Default) (Slaves: 22)Channel 23: Clear channel (Default) (Slaves: 23)Channel 24: Clear channel (Default) (Slaves: 24)Channel 25: Clear channel (Default) (Slaves: 25)Channel 26: Clear channel (Default) (Slaves: 26)Channel 27: Clear channel (Default) (Slaves: 27)Channel 28: Clear channel (Default) (Slaves: 28)Channel 29: Clear channel (Default) (Slaves: 29)Channel 30: Clear channel (Default) (Slaves: 30)Channel 31: Clear channel (Default) (Slaves: 31)
31 channels to configure.
zapata.conf
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
; ---------------- Options for use with signalling=ss7 -----------------
signalling=ss7
ss7type = itu
;ss7_called_nai=dynamic
linkset = 1
pointcode =5770           ; 5760 server B 
adjpointcode = 5760      ;5770 server B
defaultdpc = 5760         ;5770 server B
networkindicator=national
context=ss7
sigchan => 16
cicbeginswith=1
channel=>1-15
cicbeginswith=17
channel=>17-31
 
extensions.conf 
[general]
static=yes
writeprotect=no
[globals]
[default]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
include =>ss7
include =>123
[ss7]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
[123]
include =>ss7
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _XXXX,1,Dial(Zap/r1/${EXTEN})
 
when do cli asterisk at server A 
Asterisk Ready.  == Parsing '/etc/asterisk/cli.conf':   == Found*CLI> --- SS7 Up ---Resetting CICs 1 to 15Resetting CICs 17 to 31Got reset acknowledgement from CIC 1 to 15.Got reset acknowledgement from CIC 17 to 31. 
= Using SIP RTP CoS mark 5    -- Executing [1105 at 123:1] Dial("SIP/105-099c4e80", "Zap/r1/1105") in new stack    -- Called r1/1105 WARNING[3689]: app_dial.c:824 wait_for_answer: Unable to forward voice or dtmfWARNING[3689]: app_dial.c:824 wait_for_answer: Unable to forward voice or dtmf   -- Hungup 'Zap/1-1'   -- No one is available to answer at this time (1:0/0/0)   -- Auto fallthrough, channel 'SIP/105-099c4e80' status is 'NOANSWER'
server  B
 NOTICE[4160]: chan_zap.c:9696 ss7_linkset: Received RLC out and we haven't sent REL.  Ignoring.
 
thanx in advance 
ayman
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