Hello, I'm trying to make the move from Asterix, but I'm running into some difficulties. I'm try to bridge a call using our gateway however it doesn't work. In wireshark I can see I'm getting an SIP 401 Unauthorized error with a WWW-Authenticate header, after FS send the INVITE message to the gateway. However FS doesnt seem to respond to the request for Authentication. Asterix responds correctly however I cant seem to make FS to do the same. Any help would be appreciated INVITE sip:+15618911806@4.55.35.60:5070 SIP/2.0 Via: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKB49SZQHrgaaKc Max-Forwards: 8 From: "LyonL" <sip:+14142211800@207.67.30.226:5060>;tag=eZe8gcQgXXv5c To: <sip:+15618911806@4.55.35.60:5070> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f CSeq: 17931324 INVITE Contact: <sip:1-F2la9@207.67.30.226:5060;transport=udp;gw=level3> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 227 X-FS-Support: update_display Remote-Party-ID: "LyonL" <sip:+14142211800@207.67.30.226:5060>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6 s=FreeSWITCH c=IN IP4 10.126.200.6 t=0 0 m=audio 17944 RTP/AVP 0 8 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 207.67.30.226;received=207.67.30.226;branch=z9hG4bKB49SZQHrgaaKc;rport=42534 From: "LyonL" <sip:+14142211800@207.67.30.226:5060>;tag=eZe8gcQgXXv5c To: <sip:+15618911806@4.55.35.60:5070>;tag=SD6soqf99-1367649635-1316534779161 Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f CSeq: 17931324 INVITE WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks" Content-Length: 0 <include> <extension name="4142211800"> <condition field="destination_number" expression="^(\+?1)?(4142211800)$"> <action application="set" data="effective_caller_id_name=LyonL"/> <action application="set" data="effective_caller_id_number=+14142211800"/> <action application="bridge" data="sofia/gateway/level3/+15618911806"/> </condition> </extension> </include> <include> <gateway name="level3"> <param name="apply-inbound-acl" value="level3"/> <param name="username" value="1-F2la9"/> <param name="password" value="password"/>> <param name="realm" value="BroadWorks"/> <param name="proxy" value="4.55.35.60:5070"/> <param name="from-domain" value="207.67.30.226:5060"/> <param name="dtmf-type" value="rfc2833"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="register" value="false"/> </gateway> </include> <profile name="external"> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <!-- This profile is only for outbound registrations to providers --> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. --> <!-- <param name="shutdown-on-fail" value="true"/> --> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <!--<param name="enable-100rel" value="true"/>--> <!--<param name="disable-srv503" value="true"/>--> <!-- This could be set to "passive" --> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!-- used to share presence info across sofia profiles manage-presence needs to be set to passive on this profile if you want it to behave as if it were the internal profile for presence. --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!--<param name="force-register-domain" value="$${domain}"/>--> <!--all inbound reg will stored in the db using this domain --> <!--<param name="force-register-db-domain" value="$${domain}"/>--> <!-- ************************************************* --> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <!--<param name="challenge-realm" value="auto_from"/>--> <param name="user-agent-string" value="FreeSWITCH"/> <!-- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS! --> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <!--<param name="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="$${external_ssl_enable}"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) --> <param name="tls-sip-port" value="$${external_tls_port}"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile>