SIP / Caller ID / Dial

General Help
1

Hi

My Asterisk is connected to a VOIP provider through a SIP Trunk. Is working Fine. But know I need to send some calls to a remote asterisk, but I need them to receive the original Caller ID. I tried with “Dial = SIP/[email protected]” (in sip_additiona.conf).The inbound route is sending the call directly to an extension and that extension is the one with the above Dial.
I received the call but no Caller ID…

Any Suggestions?

Thanks

Leo

2

You need to learn how to create trunks between Asterisk boxes - you want to create a new IAX trunk at each box pointing at the other system. Making additions to .conf files without knowing what you are doing is pretty much guaranteed to not work for very long (but go ahead, make a bunch of changes to .conf files and see what happens, just don’t come back here asking for help when everything blows up!)

Or if you just want a shortcut to a single extension on the other box, you could do that by creating a Custom Extension, again from within FreePBX.

The point is that if you are using FreePBX you do things from within FreePBX, at least until you have a lot more familiarity with which files you can change and which you can’t. You don’t just start adding code to whichever .conf files you happen to find on the system. Try picking up one of the free e-books “Elastix Without Tears” or “PiaF without Tears” (each covers a distribution that includes FreePBX and Asterisk, so while not all the information may be relevant to your installation, much of it will be). Those will teach you the basics of working with FreePBX.

3

Do not edit the sip_additional.conf file. There is a note at the top of the file for a reason.

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

If you go to the link mentioned it will tell you what you can edit. The sip_additional.conf file is re-generated every time you apply changes so you’ll loose your edits…

4

I did not change the .conf files I did it through freepbx. I just Changed the Dial box in the extension configuration.

I need this because I need to redirect all incoming calls (from specific DID) to that remote SIP.

I’m receiving the caller ID and everything is working fine, But is this the best way to do it?

Thanks