Hi all, As H323 is not currently supported, I need to configure freeswitch to route calls to Yate2 to process my H323 calls and vice versa. For that I created the gateway in sip_profiles/external.xml <profile name="yate"> <gateways> <gateway name="192.168.1.23"> <param name="realm" value="192.168.1.23"/> <param name="extension" value="1001"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${default_codecs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="auto"/> <param name="sip-ip" value="auto"/> <param name="accept-blind-reg" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> </settings> </profile> _______________________________ Note xxx.23 is IP of yate and xxx.22 is ip of freeswitch I have also created the user in directory <include> <user id="4444" mailbox="4444"> <params> <param name="password" value="1234"/> <param name="vm-password" value="4444"/> </params> <variables> <variable name="accountcode" value="4444"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="4444"/> <variable name="effective_caller_id_number" value="4444"/> </variables> </user> </include> ______________________________________________ Now for the same freeswitch user I have created Yate Line on Yate server in accfile.conf (just like gateway in freeswitch) [MyFS] enabled=yes protocol=sip username=4444 description=FS account interval=600 formats=alaw,mulaw,gsm authname=4444 password=1234 number=4444 domain=192.168.1.22 registrar=192.168.1.22 To route calls from Yate to freeswitch I have created the folowing regexroute.conf entry of Yate server ^1001$=sip/sip:1001;line=MyFS; ;this means that calls dialed to 1001 will be routed to 1001 Freeswitch server. Then I dial from user registerd on Yate to 1001 The call is not processed of freeswitch at all, but while i put ngrep on port 5060 on server Freeswitch I found the following sequences of SIP messages: ___________________________________________ U 192.168.1.23:5060 -> 192.168.1.22:5060 INVITE sip:1001@192.168.1.22 SIP/2.0. Max-Forwards: 19. Via: SIP/2.0/UDP 192.168.1.23:5060;rport;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>. Call-ID: 1932110763@192.168.1.23. CSeq: 25 INVITE. User-Agent: YATE/2.0.0. Contact: <sip:4444@192.168.1.23:5060>. Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 264. . v=0. o=yate 1218557236 1218557236 IN IP4 192.168.1.23. s=SIP Call. c=IN IP4 192.168.1.23. t=0 0. m=audio 19990 RTP/AVP 98 8 0 101. a=rtpmap:98 ILBC/8000. a=fmtp:98 mode=30. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:30. # U 192.168.1.22:5060 -> 192.168.1.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.23:5060;rport=5060;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>. Call-ID: 1932110763@192.168.1.23. CSeq: 25 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-hacked. Content-Length: 0. . # U 192.168.1.22:5060 -> 192.168.1.23:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.23:5060;rport=5060;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>;tag=4ayXjppg3X9Fe. Call-ID: 1932110763@192.168.1.23. CSeq: 25 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary. Proxy-Authenticate: Digest realm="192.168.1.22", nonce="ebe08c45-0762-435e-9986-28064ed91b10", algorithm=MD5, qop="auth". Content-Length: 0. . # U 192.168.1.23:5060 -> 192.168.1.22:5060 ACK sip:1001@192.168.1.22 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.23:5060;rport;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>;tag=4ayXjppg3X9Fe. Call-ID: 1932110763@192.168.1.23. CSeq: 25 ACK. Max-Forwards: 19. Contact: <sip:4444@192.168.1.23:5060>. User-Agent: YATE/2.0.0. Content-Length: 0. . ________________________________________________________ Means calls are coming on freeswitch server, but even there is no process on freeswitch console. Why the calls are not processed on freeswitch, and displayed "SIP/2.0 407 Proxy Authentication Required." Is there any clue, mis-configured or any other solution, please reply. Thanks, Shehzad