[asterisk-ss7] Asterisk/libSS7 functionality question

[Rony Ron]

please find below copy of the config part that deals with the PRI ( i don't
have the alcatel's config)

the schema of the installation was :

PSTN
<--------pri-------->GSM-GW<-------pri------->***ASTERISK***<-----pri------>ALCATEL
4200

;*************** PRI-GSM-GW************
group=1
context=from-bluetower
switchtype=euroisdn
pridialplan=unknown
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
signalling=pri_cpe
channel=>1-15,17-31
;
usercallerid=yes
hidecallerid=no
restrictcid=no
callerid=asreceived
usercallingpress=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
;echocancell=yes
rxgain=0.0
txgain=0.0

;************ PRI-ALCATEL 4200 **************
group=2
context=from-alc4200
immediate=no
overlapdial=yes
signalling=pri_net
channel=>32-46,48-62

BR,

On 7/11/08, Krzysztof Drewicz <krzysztofdrewicz at gmail.com> wrote:
>
> 2008/7/11 Rony Ron <upcomingbiz at gmail.com>:
>
>> I don't know if it will help i have used digium pri card as asterisk's
>> interface to an alcatel 4400 (or 4200)
>>
>> and call transfer is working fine (since more than 1 year).
>>
>
> Kindlly  please say if you have used a PRI with QSIG or DSS-1 protocol?
> What side was master (network) and what was a slave?
>
> do you have a backup of mao from PABX and *.conf files from Asterisk? if
> so, Please forward it to me to my email, I whoul like to test it in my labo.
>
> TIA.
>
> --
> Krzysztof Drewicz
> +48 504 17 55 77
>
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