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@xxxxxxxxx> To: <asterisk-ss7 at lists.digium.com> Sent: Wednesday, September 03, 2008 1:12 AM Subject: Re: 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 ?crit : > 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 ?crit : >> >>> 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? >>> >>> >>> > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20080903/41b8f3e1/attachment.htm