Were you able to get this working? If not, get a console debug log with a siptrace and drop it on pastebin.freeswitch.org. Be sure to use "FreeSWITCH Log" as the syntax highlighting. at fs_cli you need to do: console loglevel debug sofia global siptrace on That will turn on all the debug stuff you need. From there make the call attempt and capture the output, then drop on pastebin. Give us the pb URL in this thread. Thanks, MC On Tue, Sep 20, 2011 at 9:34 AM, Chad Vogel <cvogel@lyonl.com> wrote:
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>
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