I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002 Asterisk settings: users.conf: [1002] context = DID_1002 host = *.133 username = 1002 secret = ****** trunkname = WellGate-1002 ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no host = dynamic disallow = all allow = ulaw,alaw,gsm,g726,g729 extensions.conf 1002 = SIP/1002 ... [DID_1002] exten => _88621002,1,NoOp(${CALLERID(num)}) exten => _88621002,n,Wait(1) exten => _88621002,n,SayUnixTime include = DID_1001_timeinterval_working day|${timeinterval_working day} include = DID_1001_default [DID_1001_default] exten => s,1,NoOp,${CALLERID(num)}-${CALLERID(name)} exten => s,n,Answer exten => s,n,zapateller(nocallerid) ; torture telemarketers exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,n,Hangup include = default [DID_1001_timeinterval_working day] exten = _6888,1,Goto(default|6888|1) ---- If I call in at line2, then I can hear the Time announcement and I can dial during that announcement an extension number. BTW, where can I find the additional sounds I had at an previous setup (If you know the extension, ...), which should replace the SayUnixTime I have no idea how to get dial out to work. Can anybody give me a hint, please? In Asterisk I see: [Nov 27 20:58:00] NOTICE[5095]: chan_sip.c:9227 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #102) -- Got SIP response 486 "Busy Here" back from *.133 *CLI> sip show peers 1002/1002 *.133 D 5060 Unmonitored *CLI> sip show users 1002 ****** DID_1002 No RFC3581 *CLI> sip show registry *.133:5060 1002 120 Request Sent bye Ronald _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users