> Good to here, > I know the time off set US -> AU is terrible when you need support.
I have continued to configure the analogue phone by just adding new extensions (just like any VOIP phone) to extensions.conf as follows: exten => 5162,1,SetMusicOnHold(cpwr) exten => 5162,n,Dial(Zap/32,20) exten => 5162,n,VoiceMail,5162 exten => 5162,n,Playback(vm-goodbye) exten => 5162,n,Wait(2) exten => 5162,n,HangUp() I was able to call out and call in. However, I noticed that if I dial from the analog phone to a VOIP phone, "asterisk" shows up as the dialler on the VOIP phone. This is because it is not registered in SIP. Because I think analog phone does not use SIP, so I thought I don't need to configure sip.conf. Am I correct? Did I miss anything in configuring an analog phone? _______________________________________________ -- 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
