Hi SER users iam using ser-0.9.6 when i tried with x-lite , grandstream phones , i can make calls and recieve calls now iam using one new phone it is 'ip-300' SIP phone i can register taht phone and i can make calls to that 'ip-300' but when I tried to make calls from that phone it is not success my ngrep messages are as follows : U 81.21.34.34:46117 -> 81.21.33.35:5060 INVITE sip:32331003@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9. Max-Forwards: 70. User-Agent: IP-300. From: "32331001" <sip:32331001@81.21.33.35>;tag=ib0b3FffOYuu1C8n. To: "32331003" <sip:32331003@81.21.33.35>. Call-ID: HTqDhmIJuXyigmii(a)192.168.0.74. Contact: <sip:32331001@192.168.0.74:5060>. CSeq: 1 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 289. . v=0. o=- 32362173 86500032 IN IP4 192.168.0.74. s=SIP CALL. c=IN IP4 192.168.0.74. t=0 0. m=audio 8000 RTP/AVP 18 4 0 8 3 101. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 81.21.33.35:5060 -> 81.21.34.34:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9;received= 81.21.34.34. From: "32331001" <sip:32331001@81.21.33.35>;tag=ib0b3FffOYuu1C8n. To: "32331003" <sip:32331003@81.21.33.35>;tag= 74961b5b71b6ddce908b9155b956083f.a2b9. Call-ID: HTqDhmIJuXyigmii(a)192.168.0.74. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="81.21.33.35", nonce="44ae1171fed2ef64959f27604e81bf11661bd0b5". Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=19217 req_src_ip= 81.21.34.34 req_src_port=46117 in_uri=sip:32331003@81.21.33.35 out_uri= sip:32331003@81.21.33.35 via_cnt==1". . # U 81.21.33.35:5060 -> 81.21.34.34:5060 SIP/2.0 410 Gone. Via: SIP/2.0/UDP 192.168.0.74:5060;received=81.21.34.34 ;branch=z9hG4bKZRDQYKLawgmypijq. From: <sip:32331001@81.21.33.35>;tag=0eZhJPf62anG4L1n. To: ravi <sip:32331003@81.21.33.35:5062>;tag=3806266128. Contact: <sip:32331003@81.21.34.34:5062>. Record-Route: <sip:32331001@81.21.33.35:5060;nat=yes;ftag=3806266128;lr=on>. Call-ID: 117AD09F-713F-41BF-B61F-7AFFA70FD4C5(a)192.168.0.79. CSeq: 2 INVITE. Server: X-Lite release 1105x. Content-Length: 0. . What does this mean and "this phone works with Asterisk" I think i made mistake some where any body any clues please Thank You. Regards