Hi All, When sending calls out from gateway the caller id name and number not working. Please find the freeswitch cli log and sip invite trace. Any help is appreciated. *freeswitch cli* EXECUTE sofia/sipinterface_1/barbara@alcan.mydomain.caset(effective_caller_id_name=Alcan Cable) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara@alcan.mydomain.ca SET [effective_caller_id_name]=[Alcan Cable] EXECUTE sofia/sipinterface_1/barbara@alcan.mydomain.caset(effective_caller_id_number=9052066900) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara@alcan.mydomain.ca SET [effective_caller_id_number]=[9052066900] *SIP Trace* INVITE sip:4164750401@alcan.mydomain.ca SIP/2.0 Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4 Max-Forwards: 70 Content-Length: 285 To: 4164750401 <sip:4164750401@alcan.mydomain.ca> From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f Call-ID: 43016333178f7b0882c3767aab886c1d@192.168.240.105 CSeq: 1894816474 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Barbara Ladner <sip:barbara@192.168.240.105:5060;transport=udp> Supported: replaces User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 575681134 IN IP4 192.168.240.105 s=SIP Call c=IN IP4 192.168.240.105 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.240.105:5060 ;branch=z9hG4bK17bfd0ad4;received=99.247.64.220;rport=16511 From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f To: 4164750401 <sip:4164750401@alcan.mydomain.ca> Call-ID: 43016333178f7b0882c3767aab886c1d@192.168.240.105 CSeq: 1894816474 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb Content-Length: 0 * * Thanks Lloyd