Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?)
Here is my Asterisk configuration for my incoming PSTN line: Code: [1000] type=friend host=dynamic context=incoming secret=6769 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very Inside of extensions.conf, I have this: Code: [incoming] exten => s,1,Answer( ) exten => s,2,Background(enter-ext-of-person) When I call my PSTN line, my Sipura 3000 seems to successfully answer it because the line rings once, but then immediately switches to a second dial tone. Shouldn't my incoming call be answered and then have "enter-ext-of-person" played to them? What could be causing this? Also, on a side note, I have a context called [home] which each SIP Phone is associated with. Do I need to specify each extension in there? For example: exten => 50,1,Dial(SIP/50) exten => 50,2,Hangup exten => 21,1,Dial(SIP/21) exten => 21,2,Hangup Can't I just setup a default system where any two-digit number is assumed to be an extension and it is automatically tried? Thanks for any help!! _______________________________________________ --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
