You need to setup an account for each Line button depending on how many you want eg. 2 accounts = 4 lines 3 accounts = 6 lines > max 12 lines.
Then tell Asterisk to ring at all the accounts you have setup for that phone. This works well, I had the same issue. Doug [EMAIL PROTECTED] -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jan Baggen Sent: Tuesday, November 30, 2004 11:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 7960 utilize all lines I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 -- Attempting native bridge of SIP/x.x.x.x-09a9a000 and SIP/user-acc6 -- Called user -- Got SIP response 486 "Busy here" back from x.x.x.x -- SIP/user-0e44 is busy The extensions.conf exten => s,1,Answer exten => s,2,Dial(${ARG1},300,t) exten => s,3,Ringing exten => s,4,Voicemail(u${MACRO_EXTEN}) exten => s,5,Hangup exten => s,102,Voicemail(b${MACRO_EXTEN}) exten => s,103,Hangup exten => 1234,1,Macro(oneline,SIP/user) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
