in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm
in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here [globals] [inside] exten => 77,1,voicemailmain [other] exten => 88,1,Playback(demo-congrats) Next, I have an x-lite phone set up as Display name: 40 Username: 40 Authorization user: 40 Domain/Realm: 69.240.152.95 SIP Proxy: 69.240.152.95 I get a message from SIP debug that says 40 from the x-lite is failing to register. This should be the case since I don't have any sip entry for 40. Here's the weird part. If I dial 77 from the x-lite phone I get sent to voice mail. If I dial 88 from the x-lite phone I get the demo-congrats message. Why am I getting anything? Why aren't these calls failing? _______________________________________________ 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
