Hi Damiano!
Take a look at this link:
http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159&lid=6862769263B11
Best regards;
Leonardo Kamache
On 6/5/07, damiano bertuna <damianobertuna@gmail.com>
wrote:> Hi to everybody,
>
> I have an spa 3102 where i connected an analog phone (in the fxs port) and
> the pstn line (in the fxo port).
>
> This is my problem:
>
> the incoming call doesn't arrive to asterisk.
>
> In the spa web page i configured this dialplane:
>
> (<:line01@192.168.1.220:5060>)
>
> where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip
> and 5060 is the asterisk sip port.
>
> [line01]
> username = usersipura
> fromuser = usersipura
> secret = pwdsipura
> host = 192.168.1.222
> fromdomain = 192.168.1.222
> port = 5061
> type = friend
> dtmfmode = rfc2833
> context = call_in
> insecure = very
>
>
> Why?
> is the dialplane wrong?
>
> help me, please.
>
> Damiano.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>