How to modify sip header so that asterisk will properly send a call to a Cisco CME

General Help
1

Hi all,

How would I go about modifying extensions_custom.conf to modify the To: sip header so that the extension number will pass to a Cisco CME?

Currently, when another sip phone tries to call an extension (6969 in this case) on a Cisco CME, asterisk passes this in the To: header:
To: sip:192.1.1.1:5060

What it really should be sending is this:
To: sip:[email protected]:5060

Do I need to activate the extensions_custom.conf anywhere within Freepbx?

thanks for any help!

2

Someone’s gotta know how to do this… Please help.

thx!

3

Doesn’t seem like anyone’s interested though… I’ll post what I did to make it work (changed to trunk) if anyone gives a hoot.

4

I do not see any reason for modification. Something with ip 192.1.1.1 should configured as a trunk and calls to 6969 should be routed to that trunk.

5