Hi,? I have? make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external?" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name ?Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia@192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4@sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: <document type="xml/freeswitch-httapi"><params></params><work><execute application="set" data="sip_h_api_key=b9c280dd" /><execute application="set" data="sip_h_api_secret=7678b8c4" /><execute application="set" data="sip_h_to=919818753995" /><execute application="set" data="sip_h_from=18188535351 <sip:b9c280dd@sip.nexmo.com>" /><execute application="set" data="sip_h_answer_url=http://services.qpayi.com:8080/ivr/assets/NexmoTransfer.vxml" /><dial name="exten" action="http://localhost:8080/ivr/fsdialmenu/dialendresponse/" caller-id-name="HTTAPI Test" caller-id-number="18188535351" context="default" Dialplan="XML" >919818753995</dial></work></document> here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED? Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) ?as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar?