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<div>Usually <span style="font-size:10pt">caller phone number transferring in "From:" header or in "</span><em style="font-size:small;font-style:normal;font-family:arial, sans-serif;line-height:16px;background-color:rgb(255, 255, 255)">Remote</em><span style="color:rgb(34, 34, 34);font-family:arial, sans-serif;line-height:16px;background-color:rgb(255, 255, 255);font-size:10pt">-</span><em style="font-size:small;font-style:normal;font-family:arial, sans-serif;line-height:16px;background-color:rgb(255, 255, 255)">Party</em><span style="color:rgb(34, 34, 34);font-family:arial, sans-serif;line-height:16px;background-color:rgb(255, 255, 255);font-size:10pt">-</span><em style="font-size:small;font-style:normal;font-family:arial, sans-serif;line-height:16px;background-color:rgb(255, 255, 255)">ID</em><span style="font-size:10pt">"</span></div><div><a href="http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number" target="_blank">http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number</a>
</div><div>or</div><div><a href="http://wiki.freeswitch.org/wiki/Variable_sip_cid_type" target="_blank">http://wiki.freeswitch.org/wiki/Variable_sip_cid_type</a>
</div><div><br></div><div>use it from dialplan.</div><div><br></div><div>If you still want to change "<span style="font-size:10pt"></span><span style="font-size:10pt">Contact:", try use it:</span></div><div><a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Changing_the_SIP_Contact_user" target="_blank">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Changing_the_SIP_Contact_user</a> <br><br><br><div><div id="ecxSkyDrivePlaceholder"></div>> Date: Thu, 26 Apr 2012 11:21:41 +0200<br>> From: rico-freeswitch@ricozome.net<br>> To: freeswitch-users@lists.freeswitch.org<br>> Subject: [Freeswitch-users] SIP invite contact header... again<br>> <br>> Hi list,<br>> <br>> I have some analogic phones (to be axact two Siemens Gigaset DECT<br>> phones) connected to my FS server through a RTC-to-SIP gateway.<br>> <br>> It is working well, except for announcing the caller phone number. After<br>> some investigations, it appear the SIP gateway expect the caller number<br>> into the "Contact" SIP header like this :<br>> <br>> Contact: <sip:105002@46.28.168.58:5060><br>> <br>> I found a lot of documentation related to manipulating SIP header on<br>> gateways, but in that particular case, I need to alter SIP contact<br>> header from and to a particular extension, and most of call don't even<br>> transit through a SIP gateway as they are handled into the same SIP profile.<br>> <br>> I guess this is trivial, but I'm still a newbie on IPBX and FS, and<br>> explanations I found on the wiki are too obsure for me...<br>> <br>> Thanks for your help,<br>> <br>> -- <br>> Eric<br>> <br>> _________________________________________________________________________<br>> Professional FreeSWITCH Consulting Services:<br>> consulting@freeswitch.org<br>> http://www.freeswitchsolutions.com<br>> <br>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>> http://www.cudatel.com<br>> <br>> Official FreeSWITCH Sites<br>> http://www.freeswitch.org<br>> http://wiki.freeswitch.org<br>> http://www.cluecon.com<br>> <br>> FreeSWITCH-users mailing list<br>> FreeSWITCH-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br></div></div> </div></body>
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