[Asterisk-Users] Dialing analog extensions from SIP?

[Eric "ManxPower" Wieling]

The phone's built in dialplan is prolly blocking the call.  Check the 
docs for your SIP device.  Remember SIP devices collect all digits, then 
pass them on to Asterisk as one packet.

Also what Zap port is your analog phone connected to?  What card are you 
using?

Robert La Ferla wrote:
> Doug Lytle wrote:
>> >> Is it possible to group all analog (regular phone) extensions so 
>> that you can dial it from a SIP extension?   i.e.  for use as an intercom
>> >>
>> >> I tried this:
>> >>
>> >> [default]
>> >> exten => #3001,1,Dial(Zap/1,25,t,r)
>> >> exten => #3001,2,Hangup
>>
>>
>> Change your dial to:
>>
>> exten => #3001,1,Dial(ZAP/25,tr)
>>
>> Doug
>>
> This didn't work.  I still get "Call Failed" followed by a fast busy tone.
> 
> _______________________________________________
> --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
>