Using engin, i wanted to supress caller id on all outgoing calls. This is easy, i hear you say just by prefixing numbers with 1831. For engin, this works for everywhere but an 07 number. I called them about it and they told me that its not a problem at their end. This seemed strange to me as other Australian numbers worked perfectly. Anyway, i discovered that if you use a PAP2T or a SPA-3000, and supress caller id with them, the CID sent to engin is “anonymous”.

I now knew how to supress it but had trouble getting trixbox (or FreePBX) to actually use it. I decided that I could use the “Emergency CID” field in FreePBX Extensions and only mark engin as the emergency route. Then on the outgoing routes, tick Emergency for the numbers you want to supress caller id on.

I know its a bodgy solution but it works and i would present a normal CID for emergency calls anyway.

This was annoying as I wanted more control in the inbound routes of trixbox. The following extensioins_custom.conf contect worked for me.

[custom-from-engin]
include => ext-did
exten => _X.,1,SIPGetHeader(Var_TO=To)
exten => _X.,n,Goto(ext-did,switch${Var_TO:5:10},1)

As you can see, the SIP header contains a TO header which can be looked at. I simply add a Custom app inbound route for the main number like “custom-from-engin,${EXTEN},1″ and using the above context, it re-routes the call back to the inbound contect (ext-did) with thenumber called with a “switch” prefix.

Then its just a simple matter of adding inbound routes like
switch0297990000 and making them do as you like.

using Trixbox, the + character is reserved in outbound routes so i had to use extensions_custom.conf to make the following changes…

[from-internal-intldial]
exten => _+61XXXXXXXXX,1,Goto(from-internal,0${EXTEN:-9},1)
exten => _*X.,1,Goto(from-internal,${EXTEN},1)
exten => _X.,1,Goto(from-internal,${EXTEN},1)
This allows  my saved phone entries witha + to work.

Obviously, unless you live in Australia,  you may want to change the +61 to something else.

Also, even though asterisk doesn’t appear to list + as a dialplan option, it works.

Tested in asterisk 1.2.4 and 1.2.5

While running polycom firmware up to 2.1.1 this error appears. If you change to 2.1.2, the error goes away.

UPDATE: 2007/07/10

It appears that its come back. After a particularly nasty asterisk crash, im getting these errors again.

Bugger.