[asterisk-ss7] chan_ss7 unable to match extension

[Anton]

Ususally you've got syntax misconfiguration in 
extensions.conf - see _X. or X. - behave differently.

On Wednesday 03 September 2008 13:51, Nguyen Trung Thanh 
wrote:
> Dear all,
>
>
> I am setting SS7 link using on chan_ss7. I have one link
> for signal, one link for voice. I also could make a
> outgoing call. However, I cannot receive any incoming
> call. When I make incoming call I receive:
>
> Please help me!!!!!!
>
> Tks alot!!!!
>
> console:
> --------------------------
> [Sep  3 22:34:39] NOTICE[5621]: isup.c:489
> decode_isup_phonenum: National (significant) or unknown
> nature of address indicator (1), assuming international
>
> [Sep  3 22:34:39] DEBUG[5621]: l4isup.c:2466
> process_circuit_message: Process circuit message IAM,
> CIC=5, state=0, reset_done=1 -- Recv IAM CIC=5   
> ANI=912668181 DNI=007308123 RNI= redirect=no/0 complete=0
>
> [Sep  3 22:34:39] DEBUG[5621]: l4isup.c:2644 process_iam:
> IAM cic=5, owner=0x00000000
>
> [Sep  3 22:34:39] DEBUG[5621]: l4isup.c:1530
> check_iam_sam: Unable to match extension, context: ss7,
> dni: 007308123, rni: --------------------------
>
> ss7.conf:
> --------------------------
> [linkset-siuc]
> enabled => yes
>
> enable_st => no
>
> ; Reply incoming call with CON rather than ACM and ANM
> use_connect => no
>
> hunting_policy => even_mru
>
> context => ss7
>
> language => da
>
> t35 => 15000,timeout
>
> subservice => auto
>
> [link-si]
>
> linkset => siuc
> channels =>
> schannel => 1
> firstcic =>33
> enabled => yes
> echocancel => no
> echocan_train => 350
>
> ;Span for voice
> [link-vo]
> linkset => siuc
> channels => 1-31
> schannel =>
> firstcic => 1
> enabled => yes
> echocancel => no
> echocan_train => 350
> echocan_taps => 128
>
> [host-gw1.enum.cdit.com.vn]
> enabled => yes
> opc => 0x11ae
> dpc => siuc:0x11a9
> links => si:4,vo:2
> ssn => 7
> --------------------------
>
> zaptel.conf
> ---------------------------------------
> span=1,0,0,ccs,hdb3,crc4
>
> #span=2,1,0,ccs,hdb3,crc4
> span=2,1,0,ccs,hdb3,crc4
>
> span=3,0,0,ccs,hdb3,crc4
> span=4,0,0,ccs,hdb3,crc4
>
> #span 1
> bchan=1-15
> dchan=16
> bchan=17-31
>
> #span 2
> bchan=32-46
> dchan=47
> bchan=48-62
>
> #span 3
> bchan=63-77
> dchan=78
> bchan=79-93
>
> #span 4
> bchan=94-108
> dchan=109
> bchan=110-124
> ---------------------------------------
>
> extensions.conf
> ---------------------------------------
> [ss7]
> exten => s,1,NoOP(Called: ${EXTEN})
> exten => s,n,Answer()
> exten => s,n,Playback(hello-world)
> exten => s,n,Hangup()
> ---------------------------------------
>
>
> Nguyen Trung Thanh
>
>
> ----- Original Message -----
> From: "Rony Ron" <upcomingbiz at gmail.com>
> To: <asterisk-ss7 at lists.digium.com>
> Sent: Wednesday, September 03, 2008 1:12 AM
> Subject: Re: [asterisk-ss7] Asterisk SS7 as a STP for
> Number Portability GW
>
>
> Hi Joseph,
> your solution is very elegant,
> what are those parameters:
>
> _SS7_LSPI_IDENT=ON
> _SS7_RLT_ON=YES
>
> ?
>
> regards
>
> Joseph a écrit :
> > On 09/02/08, Rony Ron wrote:
> >> Hi,
> >> imho you can do it with call forward,
> >> you receive the number
> >> you check the database if the number is there
> >> then forward to the new number (prefixing it with what
> >> ever you want) BR,
> >
> > There is a way to redirect your call back to the
> > central(Ericsson AXE) instead of keeping the media in
> > your path.
> >
> > Here is a sample:
> >
> > exten => _X.,1,Set(_SS7_LSPI_IDENT=ON)
> > exten => _X.,n,Set(_SS7_RLT_ON=YES)
> > exten => _X.,n,Answer()
> > exten => _X.,n,Playback(demo-congrats)
> >
> >    <Do your database lookup here and than redirect the
> > call back to the ss7 switch based on your lookup
> > results and drop out of the media path>
> >
> > exten => _X.,n,Dial(ZAP/r2/8005551212,30)
> > exten => _X.,n,Hangup()
> >
> >> Virmones Pereira a écrit :
> >>> Hi,
> >>>
> >>> I would like to use asterisk with SS7 as a STP for
> >>> Number Portability GW, the idea of the system is
> >>> follow:
> >>>
> >>> When the SS7 central(Ericsson AXE) receive the call
> >>> this should be route to the Asterisk to trigger the
> >>> number portability database by SS7/ISUP method if the
> >>> asterisk found this destination number in the number
> >>> portability database asterisk will insert the Routing
> >>> Number in the begin of the called number and then
> >>> route back this call to the SS7 central.
> >>>
> >>> Ex:
> >>>
> >>> user dial 551132323232 this call go the asterisk and
> >>> asterisk turn back with 55112551132323232.
> >>>
> >>> I wanna do this operation using asterisk as a STP
> >>> where the SS7 use only the signaling channel, the
> >>> media should go directly to the SSP
> >>>
> >>> somebody knows how to do it?
> >
> > -------------------------------------------------------
> >-----------------
> >
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