[asterisk-ss7] (no subject)

[aymen warfalli]

Hi list  i think i 've sent this email by mistake I am planning  to connect two asterisk box using libss7 ,I ‘ve read the list messages ( thanks for this great job) , I installed all the packages with digium  single E1 link in both boxes with centos 5 and every thing is looking ok except when I am trying to call using sip to zap it shows some problems here is my configurations file   server A--B zaptel.confspan=1,0,0,ccs,hdb3  ;span=1,1,0,ccs,hdb3  server B bchan=1-15,17-31  dchan=16loadzone = usdefaultzone = usztcfg -vvZaptel 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=yescallwaiting=yesusecallingpres=yescallwaitingcallerid=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesgroup=1callgroup=1pickupgroup=1; ---------------- Options for use with signalling=ss7 -----------------signalling=ss7ss7type = itu;ss7_called_nai=dynamiclinkset = 1pointcode =5770           ; 5760 server B adjpointcode = 5760      ;5770 server Bdefaultdpc = 5760         ;5770 server Bnetworkindicator=nationalcontext=ss7sigchan => 16cicbeginswith=1channel=>1-15cicbeginswith=17channel=>17-31 extensions.conf [general]static=yeswriteprotect=no[globals][default]exten => s,1,Answer()exten => s,2,Playback(hello-world)exten => s,3,hangup()include =>ss7include =>123[ss7]exten => s,1,Answer()exten => s,2,Playback(hello-world)exten => s,3,hangup()[123]include =>ss7exten => _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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