so. i cant choose which number i want to be seen when i make a call, but on the other hand, i can choose which trunk i will use to make a call.
and could you tell me how i choose the trunk i want to use.?
The problem i had is solved and ofcourse it is something that anyone can set up through freepbx. Yesterday i had many comments from someone who insisted -actually was possitive that my issue cant be solved. I got some help from an IT and for sure you CAN choose which number you want to be seen when performing a call.
So now:
I ll give the simple steps for the setup through the interface to resolve this issue:
step1:
In Trunks you have to insert in maximum chanells field (of the isdn ) : 1
In pstn the maximum chanells field stays blank.
Step 2: Connectivity > outbound routs:
and you set up the Trunk Sequence for Matched Routes.
no 1: Isdn 1
no2 : isdn 2 etc
and finally : pstn
In this way, when you perform an outbound call, the head number of the isdn will allways appear in the screen of the person you dial and not the only number of the pstn.
That person is correct, you did not change you caller ID. You simply changed what ISDN channel call goes out on. You did not ask us how to select a line for outbound.
You still are limited to the caller-ID’s associated with the ISDN service and bonded to channels by your SPID’s
never asked anything on how to set up a CID.
this is what i asked:
"questions:
How i’ll be able to choose when i will dial from the pstn number (lets say adding a (9)in front of the calling number) ,
and secondly, how can i set the outgoing number to be one of the two analog isdn numbers?" (and not the pstn- obvious).
Anyways, i searched for the answer and i provided it to the forum as i ought to.
If you are going to turn this post into a clarification of what i asked and which were the wrong or not answers, i m not planning to follow.
So to clarify, you wanted to define the priority for outbound routing through your analog lines?
Alternatively, most SIP trunking services allow you to set caller ID to any number you want, and don’t have a limit on number of calls. Get rid of those RJ12s and move on up in the world to SIP!