[asterisk-ss7] chan_ss7 unable to match extension

[Nguyen Trung Thanh]

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?
>>>
>>>                                           
>>>       
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> --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