I have a server for my home equipped with a TDM400P with 1 FXO and 1 FXS.
On the server is plugged the analog line and an analog phone.
The server is accessible from the internet and the two other clients will connect with a hardphone SIP from dual NAT outside of home and by lan only.
What i want is to share only one analog line by creating simple switch over SIP.
(up to here I made all by myself)
What i want first is help about to ‘prioritize’ the analog phone.
If the line is Busy (congestion because no other channel available) i want asterisk send back to the analog phone the appropriate ring tone and not the message.
I want by this action to play a message to only the person connected by SIP (which is using the phone line to make a call) and not to the called party.
This message will tell the connected client “X want to use the phone” every 15 secondes and up to 45sec because at 60 sec asterisk will hangup both. (I have allready this message.gsm and all the following message in sound/custom)
The analog client will wait 1min and retry the call and succeed. By the action of succeeding the call, right after this event i want to set a variable that expire at the end of the call and this variable will trigger a sound message to the other client that want to connect: “X is on the phone.”
I want the same variable for the two other clients (during their calls).
However i tryed to configure (with this) a MOH but didn’t succeed ( MOH do not work for the client who is waiting for an available analog line) i hope that is not related with what i want to do.
(If you are curious:) Why? Because my french isp offer vrij calls to french cellular only from the analog line and i would like to use the service with my close familly members “En bon père de famille” as they say in the Sales Conditions (Furthermore I must use dahdi which is very close to daddy :mrgreen: ). Everybody knows the use of everybody about cellular call needs that why I want messages if the line is congested. We will use it in moderation offcourse !
Since 2007, I use asterisk and upgraded from 1.2.X to 1.6.2.x but about these messages I don’t know how to do…
Thanks for your help and your time.