Your called party number is 1105. Is that really correct? In any case, it looks like your linkset is aligned and has come up. You're getting a REL message (A hangup) with cause code 16 back in response to your IAM. If you don't know what is wrong from this information, you're going to have to talk with whoever you are peering with to figure why it doesn't like it. My suspicion is it's your destination number that you're dialing out to.
Matthew Fredrickson
----- "aymen warfalli" <awerflli at hotmail.com> wrote:
> 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.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--r
> Echo Canceller: MG2
> Configuration
> ======================
> 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 15
> Resetting CICs 17 to 31
> Got 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 dtmf
> WARNING[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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