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 <[EMAIL PROTECTED]> 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: (<:[EMAIL PROTECTED]: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
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