[asterisk-users] FXS channel banks

[Lee, John (Sydney)]

> 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?