[Freeswitch-users] SIP invite contact header... again

[rico-freeswitch at ricozome.net]

Hi list,

I have some analogic phones (to be axact two Siemens Gigaset DECT
phones) connected to my FS server through a RTC-to-SIP gateway.

It is working well, except for announcing the caller phone number. After
some investigations, it appear the SIP gateway expect the caller number
into the "Contact" SIP header like this :

Contact: <sip:105002 at 46.28.168.58:5060>

I found a lot of documentation related to manipulating SIP header on
gateways, but in that particular case, I need to alter SIP contact
header from and to a particular extension, and most of call don't even
transit through a SIP gateway as they are handled into the same SIP profile.

I guess this is trivial, but I'm still a newbie on IPBX and FS, and
explanations I found on the wiki are too obsure for me...

Thanks for your help,

-- 
Eric