hi this is the linkstate capture from cli
Server A
*CLI> ss7 show linkset 1
SS7 linkset 1 status: Up
*CLI> ss7 debug linkset 1
Enabled debugging on linkset 1
*CLI> Len = 3 [ 83 84 00 ]
FSN: 4 FIB 1
BSN: 3 BIB 1
>[0] FISU
== Using SIP RTP CoS mark 5
-- Executing [1105 at 123:1] Dial("SIP/105-089a3180", "Zap/r1/1105") in new stack
-- Called r1/1105
Len = 31 [ 83 85 1c 85 80 96 a2 15 01 00 01 00 60 01 0a 00 02 07 05 83 10 11 50 0f 0a 04 83 10 01 05 00 ]
FSN: 5 FIB 1
BSN: 3 BIB 1
>[0] MSU
[ 83 85 1c ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5770 DPC 5760 SLS 1
[ 80 96 a2 15 ]
CIC: 1
[ 01 00 ]
Message Type: IAM
[ 01 ]
--FIXED LENGTH PARMS[4]--
Nature of Connection Indicator:
Satellites in connection: 0
Continuity Check: Check not required (0)
Outgoing half echo control device: not included (0)
[ 00 ]
Forward Call Indicators:
Nat/Intl Call Ind: call to be treated as a national call (0)
End to End Method Ind: no end-to-end method(s) available (0)
Interworking Ind: no interworking encountered (0)
End to End Info Ind: no end-to-end information available (0)
ISDN User Part Ind: ISDN user part used all the way (1)
ISDN User Part Pref Ind: ISDN user part not preferred all the way (1)
ISDN Access Ind: originating access ISDN (1)
SCCP Method Ind: no indication (0)
[ 60 01 ]
Calling Party Category:
Category: Ordinary calling subscriber (10)
[ 0a ]
Transmission Medium Requirements:
Speech (0)
[ 00 ]
--VARIABLE LENGTH PARMS[1]--
Called Party Number:
Nature of address: 3
NI: 0
Numbering plan: 1
Address signals: 1105#
[ 05 83 10 11 50 0f ]
--OPTIONAL PARMS--
Calling Party Number:
Nature of address: 3
NI: 0
Numbering plan: 1
Presentation: 0
Screening: 0
Address signals: 105
[ 0a 04 83 10 01 05 ]
Len = 16 [ 85 84 0d 85 8a 16 a0 15 01 00 0c 02 00 02 81 90 ]
FSN: 4 FIB 1
BSN: 5 BIB 1
<[0] MSU
[ 85 84 0d ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5760 DPC 5770 SLS 1
[ 8a 16 a0 15 ]
CIC: 1
[ 01 00 ]
Message Type: REL
[ 0c ]
--VARIABLE LENGTH PARMS[1]--
Cause Indicator:
Coding Standard: 0
Location: 1
Cause Class: 1
Cause Subclass: 0
Cause: Normal call clearing (16)
[ 02 81 90 ]
Len = 12 [ 84 86 09 85 80 96 a2 15 01 00 10 00 ]
FSN: 6 FIB 1
BSN: 4 BIB 1
>[0] MSU
[ 84 86 09 ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5770 DPC 5760 SLS 1
[ 80 96 a2 15 ]
CIC: 1
[ 01 00 ]
Message Type: RLC
[ 10 ]
WARNING[4250]: 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/105-089a3180' status is 'NOANSWER'
serve B
ss7 debug linkset 1
Enabled debugging on linkset 1
*CLI> Len = 3 [ 84 83 00 ]
FSN: 3 FIB 1
BSN: 4 BIB 1
>[0] FISU
Len = 31 [ 83 85 1c 85 80 96 a2 15 01 00 01 00 60 01 0a 00 02 07 05 83 10 11 50 0f 0a 04 83 10 01 05 00 ]
FSN: 5 FIB 1
BSN: 3 BIB 1
<[0] MSU
[ 83 85 1c ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5770 DPC 5760 SLS 1
[ 80 96 a2 15 ]
CIC: 1
[ 01 00 ]
Message Type: IAM
[ 01 ]
--FIXED LENGTH PARMS[4]--
Nature of Connection Indicator:
Satellites in connection: 0
Continuity Check: Check not required (0)
Outgoing half echo control device: not included (0)
[ 00 ]
Forward Call Indicators:
Nat/Intl Call Ind: call to be treated as a national call (0)
End to End Method Ind: no end-to-end method(s) available (0)
Interworking Ind: no interworking encountered (0)
End to End Info Ind: no end-to-end information available (0)
ISDN User Part Ind: ISDN user part used all the way (1)
ISDN User Part Pref Ind: ISDN user part not preferred all the way (1)
ISDN Access Ind: originating access ISDN (1)
SCCP Method Ind: no indication (0)
[ 60 01 ]
Calling Party Category:
Category: Ordinary calling subscriber (10)
[ 0a ]
Transmission Medium Requirements:
Speech (0)
[ 00 ]
--VARIABLE LENGTH PARMS[1]--
Called Party Number:
Nature of address: 3
NI: 0
Numbering plan: 1
Address signals: 1105#
[ 05 83 10 11 50 0f ]
--OPTIONAL PARMS--
Calling Party Number:
Nature of address: 3
NI: 0
Numbering plan: 1
Presentation: 0
Screening: 0
Address signals: 105
[ 0a 04 83 10 01 05 ]
Len = 16 [ 85 84 0d 85 8a 16 a0 15 01 00 0c 02 00 02 81 90 ]
FSN: 4 FIB 1
BSN: 5 BIB 1
>[0] MSU
[ 85 84 0d ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5760 DPC 5770 SLS 1
[ 8a 16 a0 15 ]
CIC: 1
[ 01 00 ]
Message Type: REL
[ 0c ]
--VARIABLE LENGTH PARMS[1]--
Cause Indicator:
Coding Standard: 0
Location: 1
Cause Class: 1
Cause Subclass: 0
Cause: Normal call clearing (16)
[ 02 81 90 ]
Len = 12 [ 84 86 09 85 80 96 a2 15 01 00 10 00 ]
FSN: 6 FIB 1
BSN: 4 BIB 1
<[0] MSU
[ 84 86 09 ]
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 5770 DPC 5760 SLS 1
[ 80 96 a2 15 ]
CIC: 1
[ 01 00 ]
Message Type: RLC
[ 10 ]
[Mar 26 01:55:48] NOTICE[4633]: chan_zap.c:9696 ss7_linkset: Received RLC out and we haven't sent REL. Ignoring.
============================================================================================= 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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