Usually caller phone number transferring in "From:" header or in "Remote-Party-ID"http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number orhttp://wiki.freeswitch.org/wiki/Variable_sip_cid_type use it from dialplan. If you still want to change "Contact:", try use it:http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Changing_the_SIP_Con...
Date: Thu, 26 Apr 2012 11:21:41 +0200 From: rico-freeswitch@ricozome.net To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SIP invite contact header... again
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@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
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