I think you can set callerid's in zaptel.conf for each analog port - I did that for a client a while ago. (from memory)
PaulH On Wed, 2008-03-26 at 15:53 +1100, Lee, John (Sydney) wrote: > > 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 _______________________________________________ -- 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
