[Freeswitch-users] SIP invite contact header... again
nasida at live.ru
Sat Apr 28 00:04:38 MSD 2012
Usually caller phone number transferring in "From:" header or in "Remote-Party-ID"http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
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_Contact_user
> Date: Thu, 26 Apr 2012 11:21:41 +0200
> From: rico-freeswitch at ricozome.net
> To: freeswitch-users at 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 at 188.8.131.52:5060>
> 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,
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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